ALE Application Partner Program Inter-Working Report

Transcription

1 ALE Application Partner Program Inter-Working Report Partner: Polycom Application type: video solutions Application name: Polycom RealPresence Trio Alcatel-Lucent Enterprise Platform: OmniPCX Enterprise The product and release listed have been tested with the Alcatel-Lucent Enterprise Communication Platform and the release specified hereinafter. The tests concern only the inter-working between the AAPP member s product and the Alcatel-Lucent Enterprise Communication Platform. The inter-working report is valid until the AAPP member s product issues a new major release of such product (incorporating new features or functionality), or until ALE International issues a new major release of such Alcatel-Lucent Enterprise product (incorporating new features or functionalities), whichever first occurs. ALE INTERNATIONAL MAKES NO REPRESENTATIONS, WARRANTIES OR CONDITIONS WITH RESPECT TO THE APPLICATION PARTNER PRODUCT. WITHOUT LIMITING THE GENERALITY OF THE FOREGOING, ALE INTERNATIONAL HEREBY EXPRESSLY DISCLAIMS ANY AND ALL REPRESENTATIONS, WARRANTIES OR CONDITIONS OF ANY NATURE WHATSOEVER AS TO THE AAPP MEMBER S PRODUCT INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF MERCHANTABILITY, NON INFRINGEMENT OR FITNESS FOR A PARTICULAR PURPOSE AND ALE INTERNATIONAL FURTHER SHALL HAVE NO LIABILITY TO AAPP MEMBER OR ANY OTHER PARTY ARISING FROM OR RELATED IN ANY MANNER TO THIS CERTIFICATE. ALE Application Partner Program Inter-working report - Edition 1 - page 1/95

2 Certification overview Date of the certification September 2016 ALE International representative AAPP member representative Alcatel-Lucent Enterprise Communication Platform Alcatel-Lucent Enterprise Communication Platform release AAPP member application release Application Category Benoit Trinité Joe Reventas OmniPCX Enterprise OXE R (L A) DMA 7000 R6.3.2 Platform Director RMX R8.6.4 HDX 7000 R Group 500 R6.0.1 RealPresence Trio 8800 R5.4.4 RealPresence Trio Visual+ R5.4.4 Conferencing (audio & video) Collaboration & UC Author(s): Reviewer(s): Benoit Trinié Rachid Himmi Revision History Edition 1: creation of the document Sept 2016 results Passed Refused Postponed Passed with restrictions Refer to the section 6 for a summary of the test results. IWR validity extension None ALE Application Partner Program Inter-working report - Edition 1 - page 2/95

3 AAPP Member Contact Information Contact name: Title: Address: Joe Reventas Director, Global Alliances 6001 America Center Drive Zip Code: City: San Jose, CA Country: USA Phone: +1 (303) Fax: Mobile Phone: +1 (303) Web site: address: ALE Application Partner Program Inter-working report - Edition 1 - page 3/95

4 TABLE OF CONTENTS 1 INTRODUCTION VALIDITY OF THE INTERWORKING REPORT LIMITS OF THE TECHNICAL SUPPORT CASE OF ADDITIONAL THIRD PARTY APPLICATIONS APPLICATION INFORMATION TEST ENVIRONMENT CONFIGURATION SUMMARY OF TEST RESULTS SUMMARY OF MAIN FUNCTIONS SUPPORTED S SCREENSHOT EXAMPLE Continuous Presence / 4 participants / Layout two by two Continuous Presence / 4 participants / Layout one and three vertical mode Continuous Presence / 4 participants / Layout one and three vertical mode full screen mode Continuous Presence / 4 participants + Screen sharing / Layout auto SUMMARY OF PROBLEMS SUMMARY OF LIMITATIONS NOTES, REMARKS TEST RESULT TEMPLATE TEST RESULTS THIRD PARTY VIDEO SYSTEM PROTOCOLS CONFIGURATION SIP REGISTRATION (OPTIONAL) GLOBAL REGISTRATION (TRUNK) OTHER SIP REGISTRATIONS (OPTIONAL, TRUNK) ENDPOINT SIP REGISTRATIONS SIP TRUNK (OPTIONAL) CAPABILITIES Basic incoming Call Basic Outgoing Call Video Escalation SIP REFER Support SIP Replaces Support DTMF Session Timer LD/SD/HD Video Quality level PCC MakeCall Automatic answer H.264 specificities observed Place an Audio call to Third party video System Place an Audio/Video call to the Third party video System Dial By Name Call-Log, Redial List CALL FROM 3RD PARTY VIDEO SYSTEM TO OXE EXT Receive an Audio call Receive an Audio/Video call Dial by Name Call-Log, Redial List MID CALL SERVICES Escalate an Audio call to Audio/Video ALE Application Partner Program Inter-working report - Edition 1 - page 4/95

5 8.8.2 Third party video System Holds/Retrieves communication Endpoint Holds/Retrieves the communication DTMF Call transfer from OXE extension: Blind transfer Call transfer from OXE extension: Attended transfer Call transfer from Third party video System: transfer (blind) Call transfer from Third party video System: transfer (Other) OXE CONFERENCE party conference from IP Touch party conference from Mastered Conference Meet-me conference THIRD PARTY VIDEO SYSTEM CONFERENCE Ad-hoc conference managed by third party Audio Dial-in, then Video escalation Audio/Video Dial-out from Third party Video System Dropped from the conference by Third Party Video System TUI Menu Blast Call CONTENT SHARING REMOTE CALL CONTROL OXE ADVANCED TELEPHONY SERVICES ANONYMOUS CALL ABANDONED CALL, CALL ROUTING ERROR TONES IPV CAC OXE H.A INFORMATION FOR END USERS NETWORK IMPAIRMENT & AUDIO/VIDEO QUALITY APPENDIX A: POLYCOM TEST PLAN FOR VIDEO ENDPOINTS AND INFRASTRUCTURE CROSS REFERENCES APPENDIX B: AAPP MEMBER S APPLICATION DESCRIPTION APPENDIX C: CONFIGURATION REQUIREMENTS OF THE AAPP MEMBER S APPLICATION APPENDIX D: ALCATEL-LUCENT ENTERPRISE COMMUNICATION PLATFORM: CONFIGURATION REQUIREMENTS APPENDIX E: AAPP MEMBER S ESCALATION PROCESS APPENDIX F: AAPP PROGRAM ALCATEL-LUCENT APPLICATION PARTNER PROGRAM (AAPP) ENTERPRISE.ALCATEL-LUCENT.COM APPENDIX G: AAPP ESCALATION PROCESS INTRODUCTION ESCALATION IN CASE OF A VALID INTER-WORKING REPORT ESCALATION IN ALL OTHER CASES TECHNICAL SUPPORT ACCESS ALE Application Partner Program Inter-working report - Edition 1 - page 5/95

6 1 Introduction This document is the result of the certification tests performed between the AAPP member s application and Alcatel-Lucent Enterprise s platform. It certifies proper inter-working with the AAPP member s application. Information contained in this document is believed to be accurate and reliable at the time of printing. However, due to ongoing product improvements and revisions, ALE International cannot guarantee accuracy of printed material after the date of certification nor can it accept responsibility for errors or omissions. Updates to this document can be viewed on: - the Technical Support page of the Enterprise Business Portal ( in the Application Partner Interworking Reports corner (restricted to Business Partners) - the Application Partner portal ( with free access. ALE Application Partner Program Inter-working report - Edition 1 - page 6/95

7 2 Validity of the InterWorking Report This InterWorking report specifies the products and releases which have been certified. This inter-working report is valid unless specified until the AAPP member issues a new major release of such product (incorporating new features or functionalities), or until ALE International issues a new major release of such Alcatel-Lucent Enterprise product (incorporating new features or functionalities), whichever first occurs. A new release is identified as following: a Major Release is any x. enumerated release. Example Product 1.0 is a major product release. a Minor Release is any x.y enumerated release. Example Product 1.1 is a minor product release The validity of the InterWorking report can be extended to upper major releases, if for example the interface didn t evolve, or to other products of the same family range. Please refer to the IWR validity extension chapter at the beginning of the report. Note: The InterWorking report becomes automatically obsolete when the mentioned product releases are end of life. ALE Application Partner Program Inter-working report - Edition 1 - page 7/95

8 3 Limits of the Technical support For certified AAPP applications, Technical support will be provided within the scope of the features which have been certified in the InterWorking report. The scope is defined by the InterWorking report via the tests cases which have been performed, the conditions and the perimeter of the testing and identified limitations. All those details are documented in the IWR. The Business Partner must verify an InterWorking Report (see above Validity of the InterWorking Report) is valid and that the deployment follows all recommendations and prerequisites described in the InterWorking Report. The certification does not verify the functional achievement of the AAPP member s application as well as it does not cover load capacity checks, race conditions and generally speaking any real customer's site conditions. Any possible issue will require first to be addressed and analyzed by the AAPP member before being escalated to ALE International. Access to technical support by the Business Partner requires a valid ALE maintenance contract For details on all cases (3 rd party application certified or not, request outside the scope of this IWR, etc.), please refer to Appendix F AAPP Escalation Process. 3.1 Case of additional Third party applications In case at a customer site an additional third party application NOT provided by ALE International is included in the solution between the certified Alcatel-Lucent Enterprise and AAPP member products such as a Session Border Controller or a firewall for example, ALE International will consider that situation as to that where no IWR exists. ALE International will handle this situation accordingly (for more details, please refer to Appendix F AAPP Escalation Process ). ALE Application Partner Program Inter-working report - Edition 1 - page 8/95

9 4 Application information Application Commercial Name Application Release Interface Type Polycom HDX Polycom RealPresence Group Polycom RealPresence Trio 8800 (with and without Visual+) Polycom RealPresence Desktop for PC (Windows) Polycom RealPresence Mobile for Android Polycom RealPresence Mobile for ios SIP, H.264, BFCP Polycom DMA The Polycom RealPresence DMA call processing software engine allows users to connect regardless of protocol standard, device, network, or location making communication between employees, partners and customers simple, yet effective. Administrators can expand and offer new services by leveraging existing communication network investments through the Polycom RealPresence DMA unifying call control application. With the broadest partner support, centralizing the dial plans, provisioning, and management is simplified without complex reconfigurations or replacements of bridges or voice IP PBXs. Polycom RMX Polycom RealPresence Collaboration Server breaks down barriers to video collaboration with broad support for existing and emerging standards, protocols, applications, devices and interoperability between unified communications (UC) environments. Only Polycom RealPresence Collaboration Server platform solutions provide native integration with leading UC applications and support for IP, H.323/SIP, PSTN and ISDN, AVC and SVC, all within a single conference platform. Polycom HDX7000 The Polycom HDX 7000 series systems provides flexible, affordable HD video conferencing for high-quality communication throughout mainstream workplace environments. Expanding the utility of visual communication quickly and easily, HDX 7000 telepresence systems are ideal for education, medical, enterprise and on-demand collaboration applications. ALE Application Partner Program Inter-working report - Edition 1 - page 9/95

10 Polycom Group Series 500 The Polycom RealPresence Group 500 solution is ideal for conference rooms and other collaborative environments, from small meeting rooms to larger rooms with dual screens. Powerful video and audio performance and interactive content collaboration bring users closer together and drive meaningful conversation for geographically dispersed teams. The compact, sleek design is easily hidden out of sight, keeping your rooms clutter-free. Polycom RealPresence Trio 8800 Polycom RealPresence Trio 8800 is the first smart hub for group collaboration that transforms the iconic three-point phone into a modular voice, video and content sharing system that can fit in any team environment, large or small. The Polycom RealPresence Trio brings the most amazing voice quality to a conference phone ever. However this is a conference phone, evolved. For the first time in a conference phone, RealPresence Trio also has the ability to allow you to share concepts and images wirelessly from your PC or tablet in HD resolution and visual engagement from real-time HD videoconferencing. Exceptional audio capabilities including: Enhanced HD Voice and automatic elimination of background noise with Polycom s exclusive NoiseBlock Advanced call handing, security, and provisioning features (H.264 AVC, High Profile, Lync Basic SVC) Effortless pairing for audio over USB (wired), Bluetooth, NFC (wireless) and IP using PC-based soft clients, personal contacts or joining calendared events One-step-to-join meeting experience through calendar integration The industry s only bass reflex port in a group audio conferencing solution, preserving the ability to hear lower frequency sound and rich multi-media audio 360 microphone coverage and up to 20-foot (6-meters) microphone pickup. Easily extend the reach with optional expansion microphones Local 5-way HD Voice conference 5 LCD, color touch screen display, smooth, responsive and intuitive Secure Wi-Fi network connectivity or Gigabit Ethernet network connectivity Built-in Power over Ethernet (PoE) Business-grade HD video conferencing and content sharing with the optional RealPresence Trio Visual+ accessory Polycom RealPresence Desktop Polycom RealPresence Desktop for Windows is a powerful, enterprise-grade collaboration app that extends video communications beyond the typical conference room setting to mobile professionals. Desktop video collaboration is as simple as glancing at a presence icon and clicking the name. RealPresence Desktop combines quality, power and ease-of-use with industry-leading interoperability, and security that is both cost e ective, and highly scalable. ALE Application Partner Program Inter-working report - Edition 1 - page 10/95

11 Polycom RealPresence Mobile for Android Polycom RealPresence Mobile for Android is a powerful, enterprise-grade collaboration app that extends video communications beyond the typical conference room setting to mobile professionals with tablets and smartphones. RealPresence Mobile combines power, innovation, and quality with industry-leading interoperability, and security that is both cost effective, and highly scalable. Polycom RealPresence Mobile for IOS Polycom RealPresence Mobile for Apple ios is a powerful, enterprise-grade collaboration app that extends video communications beyond the typical conference room setting to mobile professionals with tablets and smartphones. RealPresence Mobile combines power, innovation, and quality with industry-leading interoperability, and security that is both cost effective, and highly scalable. ALE Application Partner Program Inter-working report - Edition 1 - page 11/95

12 5 environment OTC PC OXE Voice Ext. OXE Meeting Room Ext. PSTN Extension 8088 Smart Deskphone Data Center OTMS OXE N1 Polycom PD Polycom RMA Polycom RMX Polycom Group Serie 500 Polycom HDX Polycom RealPresence (PC, Mobile) Polycom Trio 8800 w/ Visual+ Lan Lan Lan/WiFi Lan Note : This is recommended to register Trio Visual + to OXE since this equipement is above all an audio equipement which has basic Video capability. ALE Application Partner Program Inter-working report - Edition 1 - page 12/95

13 5.1 Configuration Third Party o Polycom System (*1) - Polycom Clariti Plolycom Platform Director Release Polycom DMA 7000 Release 6.3.2_P1_Build_ Polycom RMX (Resource Manager) Release EMA-V Polycom Endpoints Polycom HDX 7000 Release Polycom RealPresence Group 500 Release Polycom RealPresence Trio 8800 Release Polycom RealPresence Trio Visual+ Release Polycom RealPresence Windows Polycom RealPresence Android Polycom RealPresence ios Alcatel Lucent Enterprise: o Call Server - OmniPCX Enterprise Release R11.2 (L a) - OpenTouch Multmedia Services R2.2.1 ( ) o IP Touch digital sets o 8088 Smart Deskphone - R o ISDN Interface ALE Application Partner Program Inter-working report - Edition 1 - page 13/95

14 6 Summary of test results 6.1 Summary of main functions supported Peer to peer video call 8088 Smart Deskphone Premium ISDN Interface OTC-PC Deskphone HDX 7000 (Audio/Video) (Audio) (Audio) (Audio) RP Trio 8800 (Audio/Video) (Audio) (Audio) (Audio) RPGS 500 (Audio/Video)* (Audio) (Audio) (Audio) (SIP) RP PC (Audio/Video) (Audio) (Audio) (Audio) RP Mac (Audio/Video) (Audio) (Audio) (Audio)t RP ios (Audio/Video) (Audio)t (Audio) (Audio) RP Android (Audio/Video) (Audio) (Audio) (Audio) RPGS 500 (H323) Not Supported Not Supported Not Supported Not Supported Polycom video conference Audio/Video conference provided and managed by Polycom conference 8088 Smart Deskphone Premium ISDN Interface OTC-PC Deskphone Polycom RMX (Audio/Video) (Audio) (Audio) (Audio) Content sharing provided and managed by Polycom conference 8088 Smart Deskphone Polycom <Participant video> AND <Content> stream is displayed on 8088 RMX (*) bug fix in progress ALE Application Partner Program Inter-working report - Edition 1 - page 14/95

15 s Screenshot example Continuous Presence / 4 participants / Layout two by two Continuous Presence / 4 participants / Layout one and three vertical mode ALE Application Partner Program Inter-working report - Edition 1 - page 15/95

16 6.2.3 Continuous Presence / 4 participants / Layout one and three vertical mode full screen mode Continuous Presence / 4 participants + Screen sharing / Layout auto ALE Application Partner Program Inter-working report - Edition 1 - page 16/95

17 6.3 Summary of problems Known restrictions: - AL-E & PLCM : Blind transfer (initiated from OXE or Polycom side) o OK with Trio - AL-E : No Wide Band between 8088 and Polycom solution if calls goes through OXE network (including Trio Visual + if deployed on a remote OXE) - AL-E/PLCM : No RTCP-FB mechanism to optimize packet loss recovery in case of nonmanaged network - SIP to/from H323 : use cases NOT RECOMMENDED by Polycom, SIP authentication issue if H323 call is done to an OXE extension, only workaround is to disable SIP authentication (NOT RECOMMENDED by AL-E). OK for Polycom MCU use cases. Bug fix in progress : - AL-E : P2P call between 8088 and Polycom RealPresence Group 500 : video frozen for 1-2 minutes at the beginning of the call. crqms Possible workaround but it affects quality of content sharing displayed on 8088 (more acceptable). Minor issue to investigate : - AL-E : Session timer issue if SIP Authentication is enabled on Polycom side. crqms Not blocking - AL-E : No G.729 with OTC PC running SIP Device Mode. - AL-E & PLCM : Called/Calling party identity issue (impact call-log/redial) - AL-E : When OTC-PC hold remote party, it continue to send audio. - PLCM : For some complex scenario, blocked port on Polycom side (Forward/302) - AL-E : OXE H.A., after switchover, SIP BYE is not processed E2E by OXE. 6.4 Summary of limitations Non Functional: - AL-E : OXE H.A. OK except for established call. 6.5 Notes, remarks This is recommended to register Trio Visual + to OXE since this equipement is above all an audio equipement which has basic Video capability. Not yet ed - OXE ABC-F Network ALE Application Partner Program Inter-working report - Edition 1 - page 17/95

18 7 Result Template The results are presented as indicated in the example below: Case Id 1 case 1 Action Expected result case 2 Action Expected result case 3 Action Expected result case 4 Action Expected result The application waits for PBX timer or phone set hangs up Relevant only if the CTI interface is a direct CSTA link No indication, no error message Case Id: a feature testing may comprise multiple steps depending on its complexity. Each step has to be completed successfully in order to conform to the test. Case: describes the test case with the detail of the main steps to be executed the and the expected result N/A: when checked, means the test case is not applicable in the scope of the application OK: when checked, means the test case performs as expected NOK: when checked, means the test case has failed. In that case, describe in the field Comment the reason for the failure and the reference number of the issue either on ALE International side or on AAPP member side Comment: to be filled in with any relevant comment. Mandatory in case a test has failed especially the reference number of the issue. ALE Application Partner Program Inter-working report - Edition 1 - page 18/95

19 8 Results 8.1 Third party Video System Protocols configuration Purpose of this set of test is to check for availability of minimum set of parameters that are needed to properly interface the Third party video System Polycom System to an OpenTouch server. Case Id 1.1 Enable SIP stack w/o registration SIP global registration and SIP Digest authentication parameters. Other SIP registrations (per room,endpoints, ) and SIP Digest authentication parameters. Not ed Not ed 1.4 SIP Authentication for incoming call 1.5 Support for UDP/TCP. 1.6 Configuration for Codecs 1.7 Configuration for DTMF (RFC2833/4733, Patload Type) Detail of configuration of the Polycom System : ALE Application Partner Program Inter-working report - Edition 1 - page 19/95

20 8.2 SIP Registration (optional) Global Registration (trunk) Purpose of this set of test is to check SIP Registration of the Third party video System to the OXE server. Case Id 2.1 First SIP Rergistration, SIP ExtGw is switched IN- Service (DNS A Record) Not ed 2.2 Contact header from received Polycom System (IP/Domain Name) Not ed 2.3 Expires header received from Polycom System Not ed 2.4 Registration refresh Not ed 2.5 Un-registration, SIP ExtGw switched ouf of service Not ed 2.6 Registration authentication Not ed 2.7 Other specific REGISTER content Not ed Details on REGISTER message: ALE Application Partner Program Inter-working report - Edition 1 - page 20/95

21 8.3 Other SIP Registrations (optional, trunk) Purpose of this set of test is to check Other SIP Registrations of the Third party video System to an OXE server (Per room, endpoints, ). Case Id 3.1 First SIP Rergistration, Registrar is updated (DNS A Record) Not ed 3.2 Contact header from received Polycom System(IP/Domain Name) Not ed 3.3 Expires header received from Polycom System Not ed 3.4 Registration refresh Not ed 3.5 Un-registration, Registrar is updated Not ed 3.6 Registration authentication Not ed 3.7 Other specific REGISTER content Not ed Details on REGISTER message: ALE Application Partner Program Inter-working report - Edition 1 - page 21/95

22 8.4 Endpoint SIP Registrations Purpose of this set of test is to check SIP Registrations of the Third party video Endpoint to an OXE server. Case Id 4.1 First SIP Rergistration, Registrar is updated (DNS A Record) 4.2 Contact header from received Polycom System(IP/Domain Name) 4.3 Expires header received from Polycom System 4.4 Registration refresh 1min before expiry 4.5 Un-registration, Registrar is updated 4.6 Registration authentication 4.7 Other specific REGISTER content Details on REGISTER message: REGISTER sip:node pqa-collab.fr.alcatel-lucent.com;transport=tcp SIP/2.0 Via: SIP/2.0/TCP ;branch=z9hG4bK1b26a7db4DBDF851 From: " " <sip: @node pqa-collab.fr.alcatellucent.com>;tag=88a618aa-51082f2d To: <sip: @node pqa-collab.fr.alcatel-lucent.com> CSeq: 2 REGISTER Call-ID: cdf327c-eb6a00ce@ Contact: <sip: @ ;transport=tcp>;methods="invite, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomRealPresenceTrio-Trio_8800-UA/ Accept-Language: en Authorization: Digest username=" ", realm="oxerealm", nonce="1ccdcf47c55efde8273ccc2dd9f19da3", qop=auth, cnonce="pntmbpct7hxwczj", nc= , uri="sip:node pqa-collab.fr.alcatel-lucent.com;transport=tcp", response="82105d42a758039b41b9aafe2f65bed4", algorithm=md5 Max-Forwards: 70 Expires: 3600 Content-Length: SIP Trunk (optional) Purpose of this set of test is to check SIP Trunk between OXE Server and Third party video System. Case Id 5.1 OPTION msg responsed, IP address 5.2 OPTION msg responded, DNS A Record 5.3 OPTION msg responded, DNS SRV Record ALE Application Partner Program Inter-working report - Edition 1 - page 22/95

23 Case Id 5.4 Failover, OPTION msg responsed, DNS A Record Not tested 5.5 Failover, OPTION msg responsed, DNS SRV Record Not tested 8.6 Capabilities Purpose of set of this set of test is to identify SIP Profile to be used with the Third Party Video System Basic incoming Call For this set of tests, we use a Polycom endpoint equipment as the remote party. Case Id An Audio/Video call is established from Third party video System Polycom System. To OXE 8088 SIP Device Headers (From, Contact, Allow, Supported, Accepted, Session-Expires, Via) sent by Polycom System SDP Offer content (codecs, feature s options) sent by Polycom System G.722, G.711, G.729, H.264 Level 2.1 (mobile), 3.1 or 4.1 No G.729 and G.722 on Mobile Call remains established after 1 minute List of identified Audio and Video capabilities (Audio codec and details, video codec and details, profiles, level, constraints, packet mode, RTPC Feedback, ) : INVITE from Trio Visual + to 8088 : INVITE sip: @node pqa-collab.fr.alcatellucent.com;user=phone;transport=tcp SIP/2.0 Via: SIP/2.0/TCP ;branch=z9hG4bK5f663e781F5BABAE From: " " <sip: @node pqa-collab.fr.alcatellucent.com>;tag=e13250b0-880c5595 To: <sip: @node pqa-collab.fr.alcatellucent.com;user=phone> CSeq: 2 INVITE Call-ID: d9cc38c0-2c9b386f-4d719125@ Contact: <sip: @ ;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomRealPresenceTrio-Trio_8800-UA/ Accept-Language: en Supported: replaces,100rel Allow-Events: conference,talk,hold Proxy-Authorization: Digest username=" ", realm="oxerealm", nonce="065b2bf7aa427c478501fe6c76ff9923", qop=auth, cnonce="wiyz46r2i8wdzks", nc= , uri="sip: @node pqa-collab.fr.alcatellucent.com;user=phone;transport=tcp", response="3d18df81a2360dc7842a8493daa90286", algorithm=md5 ALE Application Partner Program Inter-working report - Edition 1 - page 23/95

24 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 1636 v=0 o= IN IP s=polycom IP Phone c=in IP b=as:2048 t=0 0 a=sendrecv m=audio 2228 RTP/AVP a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 m=video 2230 RTP/AVP a=content:main a=label:1 a=rtpmap:100 H264/90000 a=fmtp:100 profile-level-id=640029; packetization-mode=1; maxmbps=245760; max-fs=8196 a=rtpmap:113 H264/90000 a=fmtp:113 profile-level-id=640029; packetization-mode=0; maxmbps=245760; max-fs=8196 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=428029; packetization-mode=1; maxmbps=245760; max-fs=8196 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428029; packetization-mode=0; maxmbps=245760; max-fs=8196 m=video 2232 RTP/AVP a=content:slides a=label:2 a=rtpmap:100 H264/90000 a=fmtp:100 profile-level-id=640029; packetization-mode=1; maxmbps=245760; max-fs=8196 a=rtpmap:113 H264/90000 a=fmtp:113 profile-level-id=640029; packetization-mode=0; maxmbps=245760; max-fs=8196 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=428029; packetization-mode=1; maxmbps=245760; max-fs=8196 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428029; packetization-mode=0; maxmbps=245760; max-fs=8196 m=application UDP/BFCP * a=floorctrl:c-s a=confid:1 a=userid:1 a=floorid:2 mstrm:2 a=setup:active a=connection:new INVITE from Group Serie 500 to 8088 : INVITE sip: @node pqa-collab.fr.alcatel-lucent.com SIP/2.0 Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MES SAGE,REFER,REGISTER,UPDATE ALE Application Partner Program Inter-working report - Edition 1 - page 24/95

25 From: "840002" CSeq: 3 INVITE Supported: replaces,ms-dialog-route-set-update,ms-forking,timer User-Agent: PolycomRealPresenceGroup500/6.0.1 MRD: MRE; MRC-V=1.0.1 P_Preferred_Identity: @ Content-Type: application/sdp Plcm-Call-ID: 66b63f5b-a1f3-46a ec2fe529fa87 Max-Breadth: 60 Call-ID: 70ea0d9d-bf77-4c89-ae6a-97ab4fb2fa23 Proxy-Authorization: Digest username="polycom",realm="oxerealm",nonce="9a fc973997d d28b",uri="sip: @node pqa-collab.fr.alcatellucent.com",qop=auth,nc= ,cnonce="a989c31b",response="5cc55a 828a176765a9ba9e0dc65e6be8" Via: SIP/2.0/TCP :5060;branch=z9hG4bK x Contact: <sip: :5060;transport=tcp>;proxy=replace Max-Forwards: 68 Session-Expires: 1800;refresher=uac To: <sip: @node pqa-collab.fr.alcatel-lucent.com> Content-Length: 1483 v=0 o=groupseries IN IP s=mrd=mre MRC-V=1.0.1 c=in IP b=as:6144 t=0 0 m=audio RTP/AVP a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:15 G728/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv m=video RTP/AVP b=tias: a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42801f; max-mbps=216000; max-fs=3840; sar-supported=13; sar=13 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=42801f; packetization-mode=1; maxmbps=216000; max-fs=3840; sar-supported=13; sar=13 a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=64001f; packetization-mode=1; maxmbps=216000; max-fs=3840; sar-supported=13; sar=13 a=rtpmap:96 H /90000 a=fmtp:96 CIF4=1;CIF=1;QCIF=1;SQCIF=1;CUSTOM=352,240,1;CUSTOM=704,480,1;CUSTO M=848,480,1;CUSTOM=640,368,1;CUSTOM=432,240,1 a=rtpmap:34 H263/90000 a=fmtp:34 CIF4=1;CIF=1;QCIF=1;SQCIF=1 a=sendrecv a=rtcp-fb:* ccm tmmbr a=rtcp-fb:* ccm fir m=application RTP/AVP 100 a=rtpmap:100 H224/4800 ALE Application Partner Program Inter-working report - Edition 1 - page 25/95

26 a=sendrecv m=application UDP/BFCP * a=floorctrl:c-s a=confid:1 a=userid:2 a=floorid:1 mstrm:3 a=setup:actpass a=connection:new INVITE from HDX7000 to 8088 : INVITE sip: @node pqa-collab.fr.alcatel-lucent.com SIP/2.0 Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MES SAGE,REFER,REGISTER,UPDATE Supported: timer,replaces From: "840000" <sip:840000@ >;tag=plcm_ ;epid= fcn CSeq: 3 INVITE User-Agent: PolycomHDX7000HD/ Content-Type: application/sdp Plcm-Call-ID: e22-a331-ff65ac1b92f9 Max-Breadth: 60 Call-ID: 171d1c33-f90b-4dc6-9a98-83c6885f019f Proxy-Authorization: Digest username="polycom",realm="oxerealm",nonce="9a fc973997d d28b",uri="sip: @node pqa-collab.fr.alcatellucent.com",qop=auth,nc= ,cnonce="a7e8493d",response="f7e721 4a50e b " Via: SIP/2.0/TCP :5060;branch=z9hG4bK x Contact: <sip: :5060;transport=tcp>;proxy=replace Max-Forwards: 68 Session-Expires: 1800;refresher=uac To: <sip: @node pqa-collab.fr.alcatel-lucent.com> Content-Length: 1217 v=0 o=benoit IN IP s=c=in IP b=as:1024 t=0 0 m=audio 3212 RTP/AVP a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:15 G728/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv m=video 3214 RTP/AVP b=tias: a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42801f; max-mbps=216000; max-fs=3600; sar=13 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=42801f; packetization-mode=1; maxmbps=216000; max-fs=3600; sar=13 a=rtpmap:111 H264/90000 ALE Application Partner Program Inter-working report - Edition 1 - page 26/95

27 a=fmtp:111 profile-level-id=64001f; packetization-mode=1; maxmbps=216000; max-fs=3600; sar=13 a=rtpmap:96 H /90000 a=fmtp:96 CIF4=2;CIF=1;QCIF=1;SQCIF=1 a=rtpmap:34 H263/90000 a=fmtp:34 CIF4=2;CIF=1;QCIF=1;SQCIF=1 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1;QCIF=1 a=sendrecv a=rtcp-fb:* ccm fir m=application 3216 RTP/AVP 100 a=rtpmap:100 H224/4800 a=sendrecv INVITE from RealPresence PC to 8088 : INVITE sip: @node pqa-collab.fr.alcatellucent.com;user=phone SIP/2.0 From: "ComPlatf-PAS" <sip:840006@ >;tag=5083ac9b- F3C1EB9E CSeq: 3 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,U PDATE,REFER Resource-Priority: dsn User-Agent: Polycom RealPresence Desktop for Windows ( ) Supported: 100rel,timer,replaces,resource-priority MRD: MRE; MRC-V=1.0.1 Multiplexing: EP Content-Type: application/sdp Plcm-Call-ID: d115fd7b-1d ed8-bac1f32c772d Max-Breadth: 60 Call-ID: 0d6e23ce-d31c-4e29-a7c4-5f09b3e397dc Proxy-Authorization: Digest username="polycom",realm="oxerealm",nonce="1bcc404e3ae5828be8b01e69 fd5d85bd",uri="sip: @node pqa-collab.fr.alcatellucent.com;user=phone",qop=auth,nc= ,cnonce="c2ccafec",respo nse="85228d72e6c0783a8b4420c6d " Via: SIP/2.0/TCP :5060;branch=z9hG4bK9DF1E8B0002cccf4a x Contact: <sip: :5060;transport=tcp> Max-Forwards: 68 Session-Expires: 1800;refresher=uac To: <sip: @node pqa-collab.fr.alcatel-lucent.com> Content-Length: 2169 v=0 o= IN IP s=mrd=mre MRC-V=1.0.1 c=in IP b=as:1024 t=0 0 a=sendrecv a=vnd.polycom.mba.p2p:v=1.0.1 m=audio 3230 RTP/AVP a=sendrecv a=rtpmap:118 SIRENLPR/48000/1 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:9 G722/8000 a=fmtp:9 bitrate=64000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 ALE Application Partner Program Inter-working report - Edition 1 - page 27/95

28 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:15 G728/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:119 telephone-event/8000 a=fmtp: m=video 3232 RTP/AVP a=sendrecv a=rtcp-fb:* nack pli a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr a=vnd.polycom.forcevideomode:9 a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=64001f; packetization-mode=1; maxbr=20010; sar=13 a=vnd.polycom.avcplus.p2p:111 max-temp-layer-p2p=3 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42801f; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:109 max-temp-layer-p2p=3 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=42801f; packetization-mode=1; maxbr=20010; sar=13 a=vnd.polycom.avcplus.p2p:110 max-temp-layer-p2p=3 a=rtpmap:96 H /90000 a=fmtp:96 CIF4=1;CIF=1;QCIF=1;SQCIF=1;CUSTOM=352,240,1;CUSTOM=704,480,1;CUSTO M=1024,768,1;CUSTOM=800,600,1;CUSTOM=640,480,1;T a=rtpmap:34 H263/90000 a=fmtp:34 CIF4=1;CIF=1;QCIF=1;SQCIF=1 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtpmap:106 H264-SVC/90000 a=fmtp:106 profile-level-id=56001f; packetization-mode=1; maxbr=20010; sar=13 a=rtpmap:105 H264-SVC/90000 a=fmtp:105 profile-level-id=53e01f; packetization-mode=1; maxbr=20010; sar=13 a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 m=application 3238 UDP/BFCP * a=sendrecv a=setup:actpass a=connection:new a=floorctrl:c-s m=application 3236 RTP/AVP 100 a=sendrecv a=rtpmap:100 H224/4800 INVITE from RealPresence Android to 8088 : INVITE sip: @node pqa-collab.fr.alcatellucent.com;user=phone SIP/2.0 From: <sip:840004@ >;tag=f148684d-2f67791a CSeq: 2 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,U PDATE,REFER Resource-Priority: dsn User-Agent: Polycom RealPresence Mobile - Android phone/ Supported: 100rel,timer,replaces,resource-priority MRD: MRE; MRC-V=1.0.1 Multiplexing: EP PortraitMode: EP Content-Type: application/sdp ALE Application Partner Program Inter-working report - Edition 1 - page 28/95

29 Plcm-Call-ID: 88a0ec0e ee2-a41c-f9cb20b777a0 Via: SIP/2.0/TCP :5060;branch=z9hG4bK5E81133F9467c9304d52ee Contact: <sip: :5060;transport=tcp> Max-Forwards: 69 Max-Breadth: 60 Call-ID: a77033a0-f c-8f82-4e35d569b509 Session-Expires: 1800;refresher=uac To: <sip: @node pqa-collab.fr.alcatel-lucent.com> Content-Length: 1622 v=0 o= IN IP s=mrd=mre MRC-V=1.0.1 c=in IP b=as:512 t=0 0 a=sendrecv a=vnd.polycom.mba.p2p:v=1.0.1 m=audio 3230 RTP/AVP a=sendrecv a=rtpmap:118 SIRENLPR/48000/1 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:119 telephone-event/8000 a=fmtp: m=video 3232 RTP/AVP a=sendrecv a=rtcp-fb:* nack pli a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=640015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:111 max-temp-layer-p2p=3 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=428015; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:109 max-temp-layer-p2p=3 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=428015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:110 max-temp-layer-p2p=3 a=rtpmap:106 H264-SVC/90000 a=fmtp:106 profile-level-id=560015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 a=rtpmap:105 H264-SVC/90000 a=fmtp:105 profile-level-id=53e015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 m=application 3238 UDP/BFCP * a=sendrecv a=setup:actpass a=connection:new a=floorctrl:c-s m=application 3236 RTP/AVP 100 a=sendrecv a=rtpmap:100 H224/4800 INVITE from RealPresence ios to 8088 : ALE Application Partner Program Inter-working report - Edition 1 - page 29/95

30 INVITE SIP/2.0 From: "iphone de trinite1" CSeq: 3 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,U PDATE,REFER Resource-Priority: dsn User-Agent: Polycom RealPresence Mobile - iphone ( ) Supported: 100rel,timer,replaces,resource-priority MRD: MRE; MRC-V=1.0.1 Multiplexing: EP Content-Type: application/sdp Plcm-Call-ID: 9d1be09b-718a-4de9-9bbe-467e215f7b7e Max-Breadth: 60 Call-ID: fba8a4e4-ec46-4bc0-bd af Proxy-Authorization: Digest username="polycom",realm="oxerealm",nonce="1bcc404e3ae5828be8b01e69 fd5d85bd",uri="sip: @node pqa-collab.fr.alcatellucent.com;user=phone",qop=auth,nc= ,cnonce="f01e5899",respo nse="085e9815ff9283a811d7d3f8176c3b12" Via: SIP/2.0/TCP :5060;branch=z9hG4bK40F915056f56f000d552ee x Contact: <sip: :5060;transport=tcp> Max-Forwards: 68 Session-Expires: 1800;refresher=uac To: <sip: @node pqa-collab.fr.alcatel-lucent.com> Content-Length: 1714 v=0 o= IN IP s=mrd=mre MRC-V=1.0.1 c=in IP b=as:512 t=0 0 a=sendrecv a=vnd.polycom.mba.p2p:v=1.0.1 m=audio 3230 RTP/AVP a=sendrecv a=rtpmap:118 SIRENLPR/48000/1 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:119 telephone-event/8000 a=fmtp: m=video 3232 RTP/AVP a=sendrecv a=rtcp-fb:* nack pli a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=640015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:111 max-temp-layer-p2p=3 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=428015; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:109 max-temp-layer-p2p=3 a=rtpmap:110 H264/90000 ALE Application Partner Program Inter-working report - Edition 1 - page 30/95

31 a=fmtp:110 profile-level-id=428015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:110 max-temp-layer-p2p=3 a=rtpmap:106 H264-SVC/90000 a=fmtp:106 profile-level-id=560015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 a=rtpmap:105 H264-SVC/90000 a=fmtp:105 profile-level-id=53e015; packetization-mode=1; maxmbps=36000; max-fs=1280; max-br=20010; sar=13 a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 m=application 3238 UDP/BFCP * a=sendrecv a=setup:actpass a=connection:new a=floorctrl:c-s m=application 3236 RTP/AVP 100 a=sendrecv a=rtpmap:100 H224/4800 INVITE from RMX to 8088 : INVITE sip: @node pqa-collab.fr.alcatel-lucent.com SIP/2.0 Allow: INFO,MESSAGE,SUBSCRIBE,NOTIFY,UPDATE,REFER,INVITE,ACK,OPTIONS,CANCE L,BYE P-RMX-Info: i,c, ,99,a User-Agent: Polycom/Polycom Soft MCU/8.6.4 Allow-Events: refer,conference Supported: timer,tdialog,replaces From: <sip:dma1001_ _4fe3cbf5@col.voice.aleinternational.com:5060>;tag=rmx2k_ CSeq: 3 INVITE Content-Type: application/sdp Plcm-Call-ID: 4ac6479c-f0a2-4abb-8a4c-dcccc7ce792b Max-Breadth: 60 Call-ID: 31ba81ec-343f-4f4f-a066-cb499988a8a5 Proxy-Authorization: Digest username="polycom",realm="oxerealm",nonce="9577e1c4912c42cbd535b3a5 6eb99a7c",uri="sip: @node pqa-collab.fr.alcatellucent.com",qop=auth,nc= ,cnonce="2966d593",response="fb40ab e894ca458623c1f" Via: SIP/2.0/TCP :5060;branch=z9hG4bK x Contact: <sip: :5060;transport=tcp>;isfocus Max-Forwards: 68 Session-Expires: 1800;refresher=uac To: <sip: @node pqa-collab.fr.alcatel-lucent.com> Content-Length: 1828 v=0 o= IN IP s=rmx2k Conf c=in IP b=as:1472 t=0 0 m=audio RTP/AVP a=rtpmap:118 G719/48000/2 a=fmtp:118 CBR=128000;channels=2 a=rtpmap:126 SIRENLPR/48000/2 a=fmtp:126 bitrate= a=rtpmap:125 SIREN22STEREO/48000 a=fmtp:125 bitrate= a=rtpmap:124 SIREN22STEREO/48000 a=fmtp:124 bitrate=96000 a=rtpmap:122 SIREN14STEREO/16000 a=fmtp:122 bitrate=96000 ALE Application Partner Program Inter-working report - Edition 1 - page 31/95

32 a=rtpmap:123 SIREN22STEREO/48000 a=fmtp:123 bitrate=64000 a=rtpmap:121 SIREN14STEREO/16000 a=fmtp:121 bitrate=64000 a=rtpmap:107 SIREN22/48000 a=fmtp:107 bitrate=64000 a=rtpmap:104 G7221/16000 a=fmtp:104 bitrate=32000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=24000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=16000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=maxptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp: a=vnd.polycom.plcmmaskcap:0011 a=sendrecv m=video RTP/AVP b=tias: a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=64001f; max-fs=3840; sar=13; packetization-mode=1 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=42801f; max-fs=3840; sar=13; packetization-mode=1 a=rtpmap:114 H264/90000 a=fmtp:114 profile-level-id=42801f; max-fs=3840; sar=13 a=rtpmap:108 H /90000 a=fmtp:108 CIF=1;QCIF=1;CIF4=2 a=rtpmap:34 H263/90000 a=fmtp:34 CIF=1;QCIF=1;CIF4=2 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr a=rtcp-fb:* nack pli a=sendrecv m=application RTP/AVP 100 a=rtpmap:100 H224/6400 a=sendrecv m=application UDP/BFCP * a=floorctrl:s-only a=confid:1 a=userid:2 a=floorid:1 mstrm:21 a=setup:passive a=connection:new ALE Application Partner Program Inter-working report - Edition 1 - page 32/95

33 8.6.2 Basic Outgoing Call For this set of tests, we use a 8088 equipment as the originating party (standalone device) Case Id An Audio/Video call is established from 8088 endpoint (registered as a SIP Device on OXE Node 1) to Third party video System Polycom System. INVITE successfully parsed by the Third party video System Optionnal : Third party video System Polycom System performs a SIP digest authentication. Authentication provided by OXE is accepted Ringing state : 180 Ringing provided, content (SDP or no SDP, 100rel) Call answered : 200 OK provided, Headers content (From, contact, Allow, Supported, Accepted, Session- Expires) 180 Ringing w/o SDP SDP Answer content (codecs, feature s options) Call remains established after 1 minute Audio only, 8088 needs to escalate. See next test. ALE Application Partner Program Inter-working report - Edition 1 - page 33/95

34 8.6.3 Video Escalation Case Id For this set of tests, we use a 8088 SIP Device endpoint equipment as the originating party. An Audio call established from 8088 SIP Device on Node 1 to Third party video System Polycom System escalates to Video. reinvite successfully parsed by the Third party video System Polycom System Polycom System rejects escalation (no more ports, incompatible offer), rejection is done through SDP by setting video port to Polycom System accept the video escalation Escalation answered : 200 OK provided, Headers content (From, contact, Allow, Supported, Accepted) SDP Answer content (codecs, feature s options) Video is established Endpoints: MCU : No nack/pli/sli NOK with GS500 (SIP). Video from GS500 to 8088 runs for few seconds then is frozen. Crqms Workaround : see configuration chapter Not reproduced with GS500 6.x reinvite sent by 8088 to Polycom System : INVITE sip:1001@ :5060;transport=tcp SIP/2.0 Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO Contact: <sip: @node pqa-collab.fr.alcatellucent.com;transport=tcp> Supported: replaces,timer,path,100rel User-Agent: OmniPCX Enterprise R l a Session-Expires: 1800;refresher=uac Min-SE: 900 Content-Type: application/sdp To: <sip:841001@vftcom col.voice.aleinternational.com;user=phone>;tag=4b13dc8c From: <sip: @node pqa-collab.fr.alcatellucent.com;user=phone>;tag=b8d0b77af548bf6747b9a4847e921efe Call-ID: c442d0d45b1242d3f59daeedf@ CSeq: INVITE Max-Forwards: 70 Authorization: Digest username="vftcomoxe",realm="col.voice.aleinternational.com",nonce="mtq3mzc3ota5ndg2ng==",algorithm=md5,qop=auth,cn once=" ",nc= ,uri="sip:1001@ :5060;transport =tcp",response="59ddb025e06f74a43fa05476b27ea6cd" Via: SIP/2.0/TCP ;branch=z9hG4bK627d979f9371ac46d86d47bdb5301a36 Content-Length: 466 ALE Application Partner Program Inter-working report - Edition 1 - page 34/95

35 v=0 o=oxe IN IP s=abs c=in IP t=0 0 a=sendrecv m=audio 6000 RTP/AVP c=in IP a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:101 telephone-event/8000 m=video 7000 RTP/AVP 99 a=rtpmap:99 H264/90000 a=rtcp-fb:99 nack a=rtcp-fb:99 nack pli a=rtcp-fb:99 nack sli a=fmtp:99 profile-level-id=42801f;level-asymmetryallowed=0;packetization-mode=0 a=sendrecv Answer returned by Polycom System (RMX): SIP/ OK CSeq: INVITE Call-ID: c442d0d45b1242d3f59daeedf@ From: <sip: @node pqa-collab.fr.alcatellucent.com;user=phone>;tag=b8d0b77af548bf6747b9a4847e921efe To: <sip:841001@vftcom col.voice.aleinternational.com;user=phone>;tag=4b13dc8c Via: SIP/2.0/TCP ;branch=z9hG4bK627d979f9371ac46d86d47bdb5301a36;rport=10038 Allow-Events: conference,refer,conference User-Agent: Polycom/Polycom Soft MCU/8.6.4 P-RMX-Info: i,c, ,99,a Allow: INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,UPDATE,PRACK,SUBSCRIBE,NOTIFY,BENOTIFY Require: timer Supported: tdialog,timer,100rel,ms-conf-invite,ms-early-media,plcm-ivrservice-provider,replaces,resource-priority,histinfo,ms-safe-transfer,mssender Contact: "inbound" <sip:1001@ :5060;transport=tcp>;isfocus Content-Type: application/sdp Session-Expires: 1800;refresher=uac Content-Length: 436 v=0 o= IN IP s=c=in IP b=as:64 t=0 0 m=audio RTP/AVP a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp: a=vnd.polycom.plcmmaskcap:0011 a=sendrecv m=video RTP/AVP 99 b=tias: a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801f a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr ALE Application Partner Program Inter-working report - Edition 1 - page 35/95

36 a=rtcp-fb:* nack pli a=sendrecv Answer returned by Polycom System (HDX): SIP/ OK CSeq: INVITE Call-ID: From: To: Via: SIP/2.0/TCP ;branch=z9hG4bKce ebe780a6d172daf91ba1;rport=10375 Require: timer Supported: timer Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,R EFER,REGISTER,UPDATE User-Agent: PolycomHDX7000HD/ Contact: "inbound" Content-Type: application/sdp Session-Expires: 1800;refresher=uac Content-Length: 1284 v=0 o=benoit 0 1 IN IP s=c=in IP b=as:1152 t=0 0 m=audio 3218 RTP/AVP a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:119 G7221/16000 a=fmtp:119 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:15 G728/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv m=video 3220 RTP/AVP b=tias: a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801f; max-mbps=216000; max-fs=3600; sar=13 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=42801f; packetization-mode=1; maxmbps=216000; max-fs=3600; sar=13 a=rtpmap:111 H264/90000 ALE Application Partner Program Inter-working report - Edition 1 - page 36/95

37 a=fmtp:111 profile-level-id=64001f; packetization-mode=1; maxmbps=216000; max-fs=3600; sar=13 a=rtpmap:96 H /90000 a=fmtp:96 CIF4=2;CIF=1;QCIF=1;SQCIF=1 a=rtpmap:34 H263/90000 a=fmtp:34 CIF4=2;CIF=1;QCIF=1;SQCIF=1 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 a=sendrecv Answer returned by Polycom System (Trio Visual +): SIP/ OK Via: SIP/2.0/UDP ;branch=z9hG4bK4f616a dbaff491a300185f From: sip: ;tag=f34471cc980582d7692ed df0 To: sip: :5090 CSeq: OPTIONS Call-ID: Content-Length: 548 Content-Type: application/sdp v=0 o= IN IP s=c=in IP t=0 0 m=audio 0 RTP/AVP a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp: m=video 0 RTP/AVP a=rtpmap:97 H264/90000 a=rtpmap:98 H264/90000 a=fmtp:97 profile-level-id=42800d;packetization-mode=1 a=fmtp:98 profile-level-id=42800d;packetization-mode=0 a=imageattr:* send * recv * a=rtcp-fb:* nack a=rtcp-fb:* ccm fir Answer returned by Polycom System (GS500 SIP): SIP/ OK CSeq: INVITE Call-ID: 8aacc58d9adfca d240c8cb6a6@ From: <sip: @node pqa-collab.fr.alcatellucent.com;user=phone>;tag=ff1d8f8b9671d7fffbf426fb2f17607e To: <sip:840002@vftcom col.voice.aleinternational.com;user=phone>;tag=plcm_ Via: SIP/2.0/TCP ;branch=z9hG4bK558ed e46d5aae1b5fb336553;rport=10375 Require: timer Supported: replaces,timer Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,R EFER,REGISTER,UPDATE User-Agent: PolycomRealPresenceGroup500/6.0.1 P_Preferred_Identity: @ ALE Application Partner Program Inter-working report - Edition 1 - page 37/95

38 Contact: "inbound" Content-Type: application/sdp Session-Expires: 1800;refresher=uac Content-Length: 1379 v=0 o=groupseries 0 1 IN IP s=c=in IP b=as:6144 t=0 0 m=audio RTP/AVP a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:15 G728/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv m=video RTP/AVP b=tias: a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801f; max-mbps=216000; max-fs=3840; sarsupported=13; sar=13 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=42801f; packetization-mode=1; maxmbps=216000; max-fs=3840; sar-supported=13; sar=13 a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=64001f; packetization-mode=1; maxmbps=216000; max-fs=3840; sar-supported=13; sar=13 a=rtpmap:96 H /90000 a=fmtp:96 CIF4=1;CIF=1;QCIF=1;SQCIF=1;CUSTOM=352,240,1;CUSTOM=704,480,1;CUSTOM=848, 480,1;CUSTOM=640,368,1;CUSTOM=432,240,1 a=rtpmap:34 H263/90000 a=fmtp:34 CIF4=1;CIF=1;QCIF=1;SQCIF=1 a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 a=sendrecv Answer returned by Polycom System (RealPresence PC): SIP/ OK CSeq: INVITE Call-ID: 51f7972f14442e0a07d0640ab37a5fe2@ From: <sip: @node pqa-collab.fr.alcatellucent.com;user=phone>;tag=e3f7b557e789ab5a2dfed To: <sip:840006@vftcom col.voice.aleinternational.com;user=phone>;tag= e0a8f Via: SIP/2.0/TCP ;branch=z9hG4bKef218e36be63384a2e09b1f131d74cae;rport=10375 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE, REFER MRD: MRE; MRC-V=1.0.1 Multiplexing: EP ALE Application Partner Program Inter-working report - Edition 1 - page 38/95

39 User-Agent: Polycom RealPresence Desktop for Windows ( ) Contact: "inbound" Content-Type: application/sdp Supported: timer Session-Expires: 1800;refresher=uac Require: timer Content-Length: 2029 v=0 o= IN IP s=mrd=mre MRC-V=1.0.1 c=in IP b=as:384 t=0 0 a=sendrecv a=vnd.polycom.mba.p2p:v=1.0.1 m=audio 3230 RTP/AVP a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:118 SIRENLPR/48000/1 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:9 G722/8000 a=fmtp:9 bitrate=64000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=16000 a=rtpmap:0 PCMU/8000 a=rtpmap:15 G728/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:119 telephone-event/8000 a=fmtp: m=video 3232 RTP/AVP a=sendrecv a=content:main a=label:1 a=rtcp-fb:* nack pli a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr a=vnd.polycom.forcevideomode:9 a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=64001e; packetization-mode=1; max-mbps=49000; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:111 max-temp-layer-p2p=3 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42801e; max-mbps=49000; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:109 max-temp-layer-p2p=3 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=42801e; packetization-mode=1; max-mbps=49000; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:110 max-temp-layer-p2p=3 a=rtpmap:96 H /90000 a=fmtp:96 CIF=1;QCIF=1;SQCIF=1;CUSTOM=352,240,1;T a=rtpmap:34 H263/90000 a=fmtp:34 CIF=1;QCIF=1;SQCIF=1 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtpmap:106 H264-SVC/90000 ALE Application Partner Program Inter-working report - Edition 1 - page 39/95

40 a=fmtp:106 profile-level-id=56001e; packetization-mode=1; max-mbps=49000; max-br=20010; sar=13 a=rtpmap:105 H264-SVC/90000 a=fmtp:105 profile-level-id=53e01e; packetization-mode=1; max-mbps=49000; max-br=20010; sar=13 a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 Answer returned by Polycom System (RealPresence Android): SIP/ OK CSeq: INVITE Call-ID: From: To: Via: SIP/2.0/TCP ;branch=z9hG4bK65b39e16b532cb5da27e94b0f759b8bc;rport=10375 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE, REFER MRD: MRE; MRC-V=1.0.1 Multiplexing: EP PortraitMode: EP User-Agent: Polycom RealPresence Mobile - Android phone/ Contact: "inbound" <sip:840004@ :5060;transport=tcp> Content-Type: application/sdp Supported: timer Session-Expires: 1800;refresher=uac Require: timer Content-Length: 1486 v=0 o= IN IP s=mrd=mre MRC-V=1.0.1 c=in IP b=as:384 t=0 0 a=sendrecv a=vnd.polycom.mba.p2p:v=1.0.1 m=audio 3230 RTP/AVP a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:118 SIRENLPR/48000/1 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:119 telephone-event/8000 a=fmtp: m=video 3232 RTP/AVP a=sendrecv a=content:main a=label:1 a=rtcp-fb:* nack pli a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=640015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:111 max-temp-layer-p2p=3 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=428015; max-mbps=36000; max-fs=1280; maxbr=20010; sar=13 ALE Application Partner Program Inter-working report - Edition 1 - page 40/95

41 a=vnd.polycom.avcplus.p2p:109 max-temp-layer-p2p=3 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=428015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:110 max-temp-layer-p2p=3 a=rtpmap:106 H264-SVC/90000 a=fmtp:106 profile-level-id=560015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=rtpmap:105 H264-SVC/90000 a=fmtp:105 profile-level-id=53e015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 Answer returned by Polycom System (RealPresence ios): SIP/ OK CSeq: INVITE Call-ID: From: To: Via: SIP/2.0/TCP ;branch=z9hG4bK7e1d2ab41127e434ae5f09df0c76a6b4;rport=10375 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE, REFER MRD: MRE; MRC-V=1.0.1 Multiplexing: EP User-Agent: Polycom RealPresence Mobile - iphone ( ) Contact: "inbound" <sip:840005@ :5060;transport=tcp> Content-Type: application/sdp Supported: timer Session-Expires: 1800;refresher=uac Require: timer Content-Length: 1578 v=0 o= IN IP s=mrd=mre MRC-V=1.0.1 c=in IP b=as:384 t=0 0 a=sendrecv a=vnd.polycom.mba.p2p:v=1.0.1 m=audio 3230 RTP/AVP a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:118 SIRENLPR/48000/1 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:119 telephone-event/8000 a=fmtp: m=video 3232 RTP/AVP a=sendrecv a=content:main a=label:1 a=rtcp-fb:* nack pli a=rtcp-fb:* ccm fir a=rtcp-fb:* ccm tmmbr a=rtpmap:111 H264/90000 a=fmtp:111 profile-level-id=640015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:111 max-temp-layer-p2p=3 ALE Application Partner Program Inter-working report - Edition 1 - page 41/95

42 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=428015; max-mbps=36000; max-fs=1280; maxbr=20010; sar=13 a=vnd.polycom.avcplus.p2p:109 max-temp-layer-p2p=3 a=rtpmap:110 H264/90000 a=fmtp:110 profile-level-id=428015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=vnd.polycom.avcplus.p2p:110 max-temp-layer-p2p=3 a=rtpmap:106 H264-SVC/90000 a=fmtp:106 profile-level-id=560015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=rtpmap:105 H264-SVC/90000 a=fmtp:105 profile-level-id=53e015; packetization-mode=1; max-mbps=36000; max-fs=1280; max-br=20010; sar=13 a=rtpmap:116 vnd.polycom.lpr/9000 a=fmtp:116 V=2;minPP=0;PP=150;RS=52;RP=10;PS=1400 ALE Application Partner Program Inter-working report - Edition 1 - page 42/95

43 8.6.4 SIP REFER Support For this set of tests, we use a IP Touch endpoint equipment as the remote party Case Id Place an Audio call from IP Touch 1 (from OXE node 1) to Third party video Video MCU Polycom System Capture 200OK sent by Third party video System Polycom System when answering a simple call and check for REFER in Allow header An second Audio call is placed from IP Touch 1 to IP Touch 2. While IP Touch 2 is ringing, IP Touch 2 transfer the call. A REFER is sent to Third party video MVU Polycom System, REFER is successfully parssed and accepted by Third party video System Polycom System. The full URI (including parameters) provided in Refer- To header is re-used by Third party video System Polycom System to forge the Requested URI in subsequent outgoing INVITE The SDP of the new INVITE is built based on media previously established. Third party video System Polycom System generated implicit Notify releative to the Refer request No REFER from RMX Trio : OK GS500 : NOK, Port blocked on DMA HDX: NOK, Port blocked on DMA RealP PC : NOK, Port blocked on DMA RealP And: NOK, Port blocked on DMA RealP ios: NOK, Port blocked on DMA Trio : OK Trio : NOK, Video escalation Trio ; OK Previous dialog is released at the earliest, after subsequent 180 Ringing is processe (NOTIFY Ringing) Trio : No New Call is established successfully. Trio : OK Kepp the call established for 1 mn Trio ; OK New Call is released successfully. Trio : OK Conclusion : Only Polycom System Trio does not support properly SIP REFER mechanism in OXE environment. ALE Application Partner Program Inter-working report - Edition 1 - page 43/95

44 8.6.5 SIP Replaces Support For this set of tests, we use IP Touch endpoints Case Id Place an Audio call from IP Touch 1 (from OXE node 1) to Third party video Video Polycom System Capture 200OK responded by Third party video System Polycom System when placing a simple call and check for replaces in Supported header An second Audio call is placed from IP Touch 1 to IP Touch 2. Once IP Touch 2 has answered the call, IP Touch 2 transfer the call. A REFER or INVITE with Replaces header is sent to Third party video System Polycom System, In case REFER is used, Third party video System Polycom System place an INVITE with Replaces header make reference to Replaces parameter received in REFER. In case REFER is used, the SDP Offer in subsequient INVITE from the Third party video System Polycom System is built based on media previously established New Call is established successfully Previous SIP leg is released Trio : OK GS500 : OK HDX: NOK RealP PC :NOK RealP And:NOK RealP ios: NOK Trio : OK GS500 : OKbut (1) HDX: OKbut (1) RealP PC : OKbut (2) RealP And: OKbut (2) RealP ios: OKbut (2) (1) Not really a transfer since connection prompt is heard (managed like a fresh call on Polycom side) (2) Replaces rejected (481). OXE defense to fallback on reinvite Trio : crossed release BYE/ Call remaines established for 1mn OXE Self Refered SIP Leg is supported by Polycom System Trio : OK GS500 : OK HDX: OK RealP PC : OK RealP And: OK RealP ios: OK Conclusion : Only Polycom System Trio does support properly SIP Replaces mechanism in OXE environment. ALE Application Partner Program Inter-working report - Edition 1 - page 44/95

45 ALE Application Partner Program Inter-working report - Edition 1 - page 45/95

46 8.6.6 DTMF Case Id DTMF Payloadtype can be configured on Polycom System An Audio or Audio/Video call is established from OXE to Third party video System Polycom System DTMF PT 119 (RealPresence) or 101 (all others) is used by Polycom No way to configure this PT DTMF are generated by OXE Check for PayloadType used in RTP, Perform the test with symetric DTMF payloadtype negotiation. Check for PayloadType used in RTP, Perform the test with assymetric DTMF payloadtype negotiation. An Audio call is established from Third party video System Polycom System to OXE. DTMF are generated by Third party video System Polycom System Check for PayloadType used in RTP, Perform the test with symetric DTMF payloadtype negotiation. Check for PayloadType used in RTP, Perform the test with assymetric DTMF payloadtype negotiation. ALE Application Partner Program Inter-working report - Edition 1 - page 46/95

47 8.6.7 Session Timer For this set of tests, we use a 8088 endpoint equipment as the remote party (standalone device) Case Id An Audio or Audio/Video call is established from 8088 endpoint (from OXE node 1) to Third party video System Polycom System Check for Session Timer relative headers content Wait expiration of SIP Session timers and check for session refresh. An Audio or Audio/Video call is established from IP Touch endpoint (from OXE node 1) to Third party video System Polycom System crqms If Authentication is enabled on Polycom side, OXE do not answer challenge request from Polycom on session refresh. Workaround: deactivate SIP authentication on polycom system side Check for Session Timer relative headers content Wait expiration of SIP Session timers and check for session refresh. An Audio or Audio/Video call is established from Third party video System Polycom System to a Check for Session Timer relative headers content Wait expiration of SIP Session timers and check for session refresh. An Audio or Audio/Video call is established from Third party video System Polycom System to an IP Touch (OXE Node 1). UPDATE method is used by Polycom DMA Check for Session Timer relative headers content Wait expiration of SIP Session timers and check for session refresh. UPDATE method is used by Polycom DMA ALE Application Partner Program Inter-working report - Edition 1 - page 47/95

48 8.6.8 LD/SD/HD Video Quality level Case Id Audio/Video call established between Third party video System Polycom System (Full HD) and 8088 (H.264 High Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (Full HD) and 8088 (H.264 Medium Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (Full HD) and 8088 (H.264 Low Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (HD) and 8088 (H.264 High Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (HD) and 8088 (H.264 Medium Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (HD) and 8088 (H.264 Low Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (Below HD) and 8088 (H.264 High Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (Below HD) and 8088 (H.264 Medium Profile) Call remains is this state after waiting 60 minutes Audio/Video call established between Third party video System Polycom System (Below HD) and 8088 (H.264 Low Profile) All endpoints (except RealPresence Mobile - NA)+ MCU All endpoints (except RealPresence Mobile - NA)+ MCU All endpoints (except RealPresence Mobile - NA) + MCU All endpoints (except RealPresence Mobile - NA) + MCU All endpoints (except RealPresence Mobile - NA) + MCU All endpoints (except RealPresence Mobile - NA) + MCU MCU + RealPresence PC MCU + RealPresence PC MCU + RealPresence PC MCU + RealPresence PC MCU + RealPresence PC MCU + RealPresence PC MCU + RealPresence Mobile OK but bw exceeded from 8088 MCU + RealPresence Mobile OK but bw exceeded from 8088 MCU + RealPresence Mobile OK but bw exceeded from 8088 MCU + RealPresence Mobile OK but bw exceeded from 8088 MCU + RealPresence Mobile OK but bw exceeded ALE Application Partner Program Inter-working report - Edition 1 - page 48/95

49 from Call remains is this state after waiting 60 minutes MCU + RealPresence Mobile OK but bw exceeded from 8088 ALE Application Partner Program Inter-working report - Edition 1 - page 49/95

50 PCC MakeCall But does not support Video since Ghost Z feature used in OXE does not support Video. Case Id MakeCall (INVITE/Ringing in band/reinvite) All Endpoints except H Hold/Retrieve Second Call Call Broker Conference Transfert Release Call All Endpoints except H323 All Endpoints except H323 All Endpoints except H323 A delay in audio path is noticeable from OXE to Polycom All Endpoints except H323 All Endpoints OK except H323 and RealPresence All Endpoints except H323 ALE Application Partner Program Inter-working report - Edition 1 - page 50/95

51 Automatic answer Not Applicable for SIP Device, there is however auto answer settings available on Case Id H.264 specificities observed An Audio/Video call is established from 8088 video endpoint to Third pârty Video MCU Polycom System. A network capture is performed and analysed to establish the list of capability. List of capabilities related to Video observed for Polycom System : Case Id Multiples Reference Frame supported Not ed Multiples Reference Frame used Not ed Packetiezation mode (PM0, PM1, STAP-A) Interworking limited to PM0 because of asymmetric negotiation issue on 8088 side. Which endpoint? Advertised Profile/Level in SPS in conformance to SDP negotiation Not ed Configuration option to limit Video bandwifth RTCP FB NACK PLI negotiation MCU : OK Endpoints : OK MCU :OK Endpoint HDX : OK Endpoints GS : KO Endpoints Trio : KO Endpoints RP : OK RTCP FB NACK SLI negotiation NO RTCP FB generic NACK negotiation NO RTCP FB FIR negotiation SDP parameter for Bandwidth Control Only on RealPresence AS, TIAS (except Trio Visual + : only AS) ALE Application Partner Program Inter-working report - Edition 1 - page 51/95

52 INFO based Video Fast Update (respond) INFO based Video Fast Update (request) INFO based Bit Rate modification (respond) INFO based Bit Rate modification (request) Example of SIP INFO based Video Fast Update request from Polycom System ALE Application Partner Program Inter-working report - Edition 1 - page 52/95

53 Call from OXE ext. to 3rd party Video system Place an Audio call to Third party video System Case Id The established call must be maintained at least 1 minute. Calling Number/Name is checked on third party Video System side. Connected Party Number/Name is checked on OXE side. Place an Audio call (G.722) from IP Touch (OXE Node 1) to Third party video System Polycom System Place an Audio call (G.711) from IP Touch (OXE Node 1) to Third party video System Polycom System Place an Audio call (G.729) from IP Touch (OXE Node 1) to Third party video System Polycom System Place an Audio call (G.711) from TDM Set (OXE Node 1) to Third party video System Polycom System Place an Audio call (G.722) from 8088 (OXE Node 1) to Third party video System Polycom System Place an Audio call (G.711) from PSTN Extension (OXE Node 1) to Third party video System Polycom System Place an Audio call (G.711) from 8088 (OXE Node 2) to Third party video System Polycom System Place an Audio call (G.711) from IP Touch (OXE Node 2) to Third party video System Polycom System Place an Audio call (G.729) from IP Touch (OXE Node 2) to Third party video System Polycom System Place an Audio call (G.729) from TDM Set (OXE Node 1) to Third party video System Polycom System Call is Cancelled on OXE side before the call is answered by Third party video System Polycom System Call is Rejected on Third party video System side Polycom System Check for Reason Header Call is Deflected on Third party video System side Polycom System Check for Diversion and History-Info Headers MCU + Endpoints MCU + Endpoints except H323 MCU + Endpoints OXE Limitation, no G.722 if configured in Full Call Handling mode MCU + Endpoints Not tested Not ed Not ed MCU + Endpoints OK except RealPresence Mobile SIP : 603 Decline H323 : 486 Busy Here No Reason Header ALE Application Partner Program Inter-working report - Edition 1 - page 53/95

54 Place an Audio/Video call to the Third party video System Case Id NA Video profile on Third Party Video Systel is set to default. The established call must be maintained at least 1 minute Dial By Name Case Id Place an Audio/VIdeo call from 8088 (OXE Node 1) to Third party video System Polycom System, by searching Name (LDAP Integration). Place an Audio call from IP Touch (OXE Node 1) to Third party video System Polycom System by searching Name (LDAP Integration). No H.350 compliant on 8770 side. To investigate No H.350 compliant on 8770 side Details on LDAP Integration: Call-Log, Redial List Case Id After a call has been placed from 8088 (OXE Node 1) to Third party video System Polycom System, Call- Log/Redial List is checkedd on 8088 side. After a call has been received from Third party video System Polycom System to 8088 (OXE Node 1), Call- Log/Redial List is checkedd on 8088 side. After a call has been placed from IP Touch (OXE Node 1) to Third party video System Polycom System, Call- Log/Redial List is checkedd on IP Touch side. After a call has been placed from Third party video System Polycom System to IP Touch (OXE Node 1), Call-Log/Redial List is checkedd on IP Touch side. After a call has been placed from IP Touch (OXE Node 1) to Third party video System Polycom System, Call- Log/Redial List is checked on Third party video System Polycom System. After a call has been placed from Third party video System Polycom System to IP Touch (OXE Node 1), Call-Log/Redial List is checkedd on Third party video OK in when OXE in proxy mode, NOK when OXE in full Call Handling mode But, called number not visible, only trunk ID is provided! OXE config to check OXE Config to check OK from GS500 (SIP), HDX and Trio Visual +. NOK from Trio Visual + if registered to DMA. ALE Application Partner Program Inter-working report - Edition 1 - page 54/95

55 System Polycom System. 8.7 Call from 3rd party Video system to OXE Ext Receive an Audio call Case Id Receive an Audio call (G.722) from Third party video System Polycom System to IP Touch (OXE Node 1) Receive an Audio call (G.711) from Third party video System Polycom System to IP Touch (OXE Node 1) Receive an Audio call (G.729) from Third party video System Polycom System to IP Touch (OXE Node 1) Receive an Audio call (G.711) from Third party video System Polycom System to TDM (OXE Node 1) Receive an Audio call (G.711) from Third party video System Polycom System to 8088 (OXE Node 1) Receive an Audio call (G.722) from Third party video System Polycom System to 8088 (OXE Node 1) Receive an Audio call (G.722) from Third party video System Polycom System to PSTN (OXE Node 1) Receive an Audio call (G.711) from Third party video System Polycom System to 8088 (OXE Node 2) Receive an Audio call (G.711) from Third party video endpoint Polycom System to IP Touch (OXE Node 2) Endpoints : NOK from H323 MCU : OK Endpoints : NOK from H323 MCU : OK Endpoints : NOK from H323 MCU : NOK NOK from H323 A./V Call NOK from H323 A./V Call OXE Limitation, no G.722 if configured in Full Call Handling mode Endpoints : NOK from H323 MCU :OK A./V Call Not ed Not ed Receive an Audio call (G.729) from Third party video endpoint Polycom System to IP Touch (OXE Node 2) Receive an Audio call (G.729) from Third party video endpoint Polycom System to TDM Set Receive an Audio call from Third party video System Polycom System to IP Touch (OXE Node 1) : Call is cancelled Receive an Audio call from Third party video System Polycom System to IP Touch (OXE Node 1) : Call is rejected (DND) Receive an Audio call from Third party video System Polycom System to IP Touch (OXE Node 1) : Call is deflected to another OXE extension Receive an Audio call from Third party video System Polycom System to IP Touch (OXE Node 1) : Call is forwarded to another OXE extension (immediate, no Not ed Endpoints :NOK from H323 MCU : OK Endpoints : NOK from H323 MCU : To be tested MCU + Endpoints Endpoints : NOK from H323 ALE Application Partner Program Inter-working report - Edition 1 - page 55/95

56 response, busy) MCU : NOK Receive an Audio call from Third party video System Polycom System to IP Touch (OXE Node 1) : Call is forwarded to Voic (immediate, no response, busy) Receive an Audio call from Third party video System Polycom System to IP Touch (OXE Node 1) : Call is forwarded to PSTN destination (immediate, no response, busy) Receive an Audio call (G.722) from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / SEPLOS Mode) Receive an Audio call (G.711) from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / SEPLOS Mode) Receive an Audio call (G.729) from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / SEPLOS Mode) Receive an Audio call (G.722) from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / Z/SIP Mode) Receive an Audio call (G.711) from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / Z/SIP Mode) Receive an Audio call (G.729) from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / Z/SIP Mode) Endpoints : NOK from H323 MCU : NOK Endpoints : NOK from H323 MCU : OK OXE Limitation A./V Call OXE Limitation, no G.722 Endpoints : OK MCU : OK Check OXE config (only G.711 offered to OTC PC in 3PCC) Receive an Audio/Video call Case Id The established call must be maintained at least 1 minute. Receive an Audio/VIdeo call from Third party video System Polycom System to 8088 (OXE Node 1) (8088 set in H.264 High Profile) Receive an Audio/VIdeo call from Third party video System Polycom System to 8088 (OXE Node 2) (H.264 High Profile) Receive an Audio/Video call from Third party video System Polycom System to 8088 (OXE Node 1) (8088 set in H.264 Medium Profile) Receive an Audio/Video call from Third party video System Polycom System to 8088 (OXE Node 1) (8088 set in H.264 Low Profile) Receive an Audio/VIdeo call from Third party video System Polycom System to IP Touch (OXE Node 1) Receive an Audio/VIdeo call from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / SEPLOS Mode) Receive an Audio/VIdeo call from Third party video System Polycom System to OTC PC (OTMS/OXE Node 1 / Z/SIP Mode) Sometime, with RealPresence PC, black screen on 8088 side Not ed OK, but no Video OK, but no Video ALE Application Partner Program Inter-working report - Edition 1 - page 56/95

57 8.7.3 Dial by Name Case Id Place an Audio/VIdeo to 8088 (OXE Node 1) to Third party video System Polycom System, by searching Name (Phonebook Integration). Through AD RealPresence : local directory OK Details on LDAP Integration: Call-Log, Redial List Case Id After a call has been placed Third party video System Polycom System to 8088 (OXE Node 1), Call- Log/Redial List is checkedd on Third party video System Polycom System. After a call has been placed Third party video System Polycom System to PSTN Destination, Call- Log/Redial List is checkedd on Third party video System Polycom System. ALE Application Partner Program Inter-working report - Edition 1 - page 57/95

58 8.8 Mid call services Escalate an Audio call to Audio/Video Case Id The established call must be maintained at least 1 minute after escalation takes place Place an Audio call between 8088 (OXE Node 1) and Third party video System Polycom System then escalate to Video on 8088 side (8088 set in H.264 High Profile) Place an Audio call between 8088 (OXE Node 2) and Third party video System Polycom System then escalate to Video on 8088 side (H.264 High Profile) Place an Audio call between 8088 (OXE Node 1) and Third party video System Polycom System then escalate to Video on Third party video System side (8088 set in H.264 High Profile) Place an Audio call between 8088 (OXE Node 2) and Third party video System Polycom System then escalate to Video on Third party video System side (H.264 High Profile) Not ed No way to escalate to Video on Polycom side. Not ed Third party video System Holds/Retrieves communication Case Id Audio/Video call is established between Third party video System Polycom System and 8088 (Oxe Node 1), Third party video System Polycom System puts communication on hold, 8088 is in hold state; System music is heared on 8088, Call remains is this state after waiting 1 minute HDX, MCU, RealPresence : NA Third party video System Polycom System retrieves the communication, audio and video are resumed Call remains is this state after waiting 1 minute Audio/Video call is established between Third party video System Polycom System and 8088 (Oxe Node 2), Third party video System Polycom System puts communication on hold, 8088 is in hold state; System music is heared on 8088, Not ed Call remains is this state after waiting 1 minute Not ed Third party video System Polycom System retrieves the communication, audio and video are resumed Call remains is this state after waiting 1 minute Audio only call is established between Third party video System Polycom System and IP Touch (Oxe Node 1), Third party video System Polycom System puts communication on hold, IP Touch is in hold state; System music is heared on IP Touch, ALE Application Partner Program Inter-working report - Edition 1 - page 58/95

59 Call remains is this state after waiting 1 minute Third party video System Polycom System retrieves the communication, audio only media is resumed Call remains is this state after waiting 1 minute Audio only call is established between Third party video System Polycom System and OTC PC (OTMS/Oxe Node 1/SEPLOS Mode), Third party video System Polycom System puts communication on hold, OTC PC is in hold state; System music is heared on OTC PC, Call remains is this state after waiting 1 minute Third party video System Polycom System retrieves the communication, audio only media is resumed Call remains is this state after waiting 1 minute Audio only call is established between Third party video System Polycom System and OTC PC (OTMS/Oxe Node 1/Z/SIP Mode), Third party video System Polycom System puts communication on hold, OTC PC is in hold state; System music is heared on OTC PC, Call remains is this state after waiting 1 minute Third party video System Polycom System retrieves the communication, audio only media is resumed Call remains is this state after waiting 1 minute Endpoint Holds/Retrieves the communication Case Id Audio/Video call is established between Third party video System Polycom System and 8088 (OXE Node 1), 8088 puts communication on hold, System music is heared by Third party video System Polycom System, Call remains is this state after waiting 1 minute MyIC Phone 8088 retrieves the communication, Audio/Video is resumed. Endpoints : OK except GS500 SIP (call is released - crqms ). MCU : NA Call remains is this state after waiting 1 minute ALE Application Partner Program Inter-working report - Edition 1 - page 59/95

60 8.3.5 Audio/Video call established between Third party video System Polycom System and 8088 (OXE Node 2), 8088 puts communication on hold, System music is heared by Third party video System Polycom System, Not ed Call remains is this state after waiting 1 minute Not ed MyIC Phone 8088 retrieves the communication, Audio/Video media are resumed. Not ed Call remains is this state after waiting 1 minute Not ed Audio only call established between Third party video System Polycom System and IP Touch (OXE Node 1), IP Touch puts communication on hold, System music is heared by Third party video System Polycom System, Endpoints : OK MCU : OK (but MOH can be eared in conference) Call remains is this state after waiting 1 minute MyIC Phone IP Touch retrieves the communication, Audio only media is resumed Call remains is this state after waiting 1 minute Audio only call established between OTCV user A/Third party video System Polycom System and OTC PC (OTMS/Oxe Node 1/SEPLOS Mode), OTC PC puts communication on hold, System music is heared by Third party video System Polycom System, Call remains is this state after waiting 1 minute OTC PC retrieves the communication, Audio only media is resumed Call remains is this state after waiting 1 minute Audio only call established between OTCV user A/Third party video System Polycom System and OTC PC (OTMS/Oxe Node 1/Z/SIP Mode), OTC PC puts communication on hold, System music is heared by Third party video System Polycom System, GS500 : NOK, OTC- PC in hold state continue to send voice packet, and GS500 is puzzled crqms to do Call remains is this state after waiting 1 minute OTC PC retrieves the communication, Audio only media is resumed Call remains is this state after waiting 1 minute ALE Application Partner Program Inter-working report - Edition 1 - page 60/95

61 8.8.4 DTMF Case Id An Audio or Audio/Video call is established from IP Touch to Third party video System Polycom System DTMF are sent from IP Touch to Third party video System Polycom System. DTMF are regognized by Third party video System Polycom System.. An Audio or Audio/Video call is established from OTC PC (OTMS/OXE Node 1/SEPLOS Mode) to Third party video System Polycom System DTMF are sent from OTC PC to Third party video System Polycom System. DTMF are regognized by Third party video System Polycom System.. An Audio or Audio/Video call is established from OTC PC (OTMS/OXE Node 1/Z/SIP Mode) to Third party video System Polycom System DTMF are sent from OTC PC to Third party video System Polycom System. DTMF are regognized by Third party video System Polycom System.. An Audio or Audio/Video call is established from 8088 (OXE Node 1) to Third party video System Polycom System DTMF are sent from 8088 to Third party video System Polycom System DTMF are regognized by Third party video System Polycom System.. An Audio or Audio/Video call is established from 8088 (OXE Node 2) to Third party video System Polycom System DTMF are sent from 8088 to Third party video System Polycom System. DTMF are regognized by Third party video System Polycom System.. An Audio or Audio/Video call is established from PSTN origine to Third party video System Polycom System DTMF are sent from PSTN side to Third party video System Polycom System. DTMF are regognized by Third party video System Polycom System.. An Audio or Audio/Video call is established from Third party video System Polycom System to 8088 forwarded to Voic Not ed Not ed Not ed ALE Application Partner Program Inter-working report - Edition 1 - page 61/95

62 DTMF are sent from Third party video System Polycom System to Voic application DTMF are regognized by Voic application Call transfer from OXE extension: Blind transfer Case Id The established call must be maintained at least 1 minute. Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 1) performs a blind transfer to a second 8088 (OXE Node 1). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 1) performs a blind transfer to second 8088 (OXE Node 2). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 2) performs a blind transfer to a second 8088 (OXE Node 1). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and an OTC PC (OTMS/OXE Node 1/SEPLOS Mode). OTC PC performs a blind transfer to a 8088 (OXE Node 1) answers the call. Place an Audio/Video call between Third party video System Polycom System and an OTC PC (OTMS/OXE Node 1/Z/SIP Mode). OTC PC performs a blind transfer to a 8088 (OXE Node 1) answers the call. For one of the test above, Call remains established after waiting 1 minute Endpoints : KO MCU : KO Workarround : deactivate SIP Authentication on OXE side. In this case, video is lost after transfer, need to re-escalate Not ed Not ed Not ed Endpoints : KO MCU : KO Workarround : deactivate SIP Authentication on OXE side. In this case, video is lost after transfer, need to re-escalate Call transfer from OXE extension: Attended transfer Case Id Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 1) performs an attended transfer to a second 8088 (OXE Node 1). Second 8088 answers the call. Endpoints : OK MCU : OK Video is lost, need to re-escalate. ALE Application Partner Program Inter-working report - Edition 1 - page 62/95

63 Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 1) performs an attended transfer to second 8088 (OXE Node 2). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 2) performs an attended transfer to a second 8088 (OXE Node 1). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and an OTC PC (OTMS/OXE Node 1/SEPLOS Mode). OTC PC performs an attended transfer to a 8088 (OXE Node 1) answers the call. Place an Audio/Video call between Third party video System Polycom System and an OTC PC (OTMS/OXE Node 1/Z/SIP Mode). OTC PC performs an attended transfer to a 8088 (OXE Node 1) answers the call. For one of the test above, Call remains established after waiting 1 minute Not ed Not ed Not ed Endpoints : OK except video freeze with GS500 MCU : OK Video is lost, need to re-escalate Call transfer from Third party video System: transfer (blind) Case Id Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 1). Third party video System performs a blind transfer to a second 8088 (OXE Node 1). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 1). Third party video System performs a blind transfer to second 8088 (OXE Node 2). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 2). Third party video System performs a blind transfer to a second 8088 (OXE Node 1). Second 8088 answers the call. Place an Audio/Video call between Third party video System Polycom System and an OTC PC (OTMS/OXE Node 1/SEPLOS Mode). Third party video System performs a blind transfer to a 8088 (OXE Node 1) answers the call. Place an Audio/Video call between Third party video System Polycom System and an OTC PC (OTMS/OXE Node 1/Z/SIP Mode). Third party video System performs a blind transfer to 8088 (OXE Node 1) answers the call. For one of the test above, Call remains established after waiting 1 minute Only Available on Trio Visual + : OK, audio OK, but video lost and no way to get it back Not ed Not ed No ed Call transfer from Third party video System: transfer (Other) Case Id ALE Application Partner Program Inter-working report - Edition 1 - page 63/95

64 8.8.1 Place an Audio/Video call between Third party video System Polycom System and a 8088 (OXE Node 1). Third party video System performs an attended transfer to a second 8088 (OXE Node 1). Second 8088 answers the call. Only Available on Trio Visual + : KO, audio OK, but video lost issue. 8.9 OXE Conference party conference from IP Touch Case Id Place an Audio call between IP Touch (OXE Node 1) and Third party video System Polycom System then manage a three party conference on IP Touch side Endpoints : OK MCU : OK Conference remains established after waiting 1 minute The third participant leaves the conference, P2P call between IP Touch and Third party video System Polycom System is re-established. Place an Audio call between OTC PC (OTMS/OXE Node 1/SEPLOS Mode) and Third party video System Polycom System then manage a three party conference on OTC PC side Endpoints : OK, except HDX et GS500 (duplicated call) MCU : OK Conference remains established after waiting 1 minute Place an Audio call between OTC PC (OTMS/OXE Node 1/Z/SIP Mode) and Third party video System Polycom System then manage a three party conference on OTC PC side Endpoints : OK MCU : OK Conference remains established after waiting 1 minute The third participant leaves the conference, P2P call between OTC PC (OTMS/OXE Node 1/Z/SIP Mode) and Third party video System Polycom System is reestablished party conference from 8088 Case Id NA Mastered Conference Case Id TBD Not ed ALE Application Partner Program Inter-working report - Edition 1 - page 64/95

65 8.9.4 Meet-me conference Case Id TBD Not ed 8.10 Third party Video System Conference Ad-hoc conference managed by third party Case Id Place an Audio/VIdeo call from Third party video Polycom System to 8088 (OXE Node 1). From Third party video Polycom System add a second 8088 (OXE Node 1). Check video conference behavior (active speaker, ). Place an Audio call from a 8088 (OXE Node 1) to the Third party video Polycom System. Third party video Polycom System adds a second First 8088 escalates to video. Check video conference behavior (active speaker, ). Only applicable to Trio Visual + : Only audio, no video between first 8088 and Polycom endpoint Only applicable to Trio Visual + : Only audio, no video between first 8088 and Polycom endpoint Conference remains established for 1 hour Third party video Polycom System leave the conference, conference remains established for other participants (8088). Remaining participants keep communication for 1 hour. Third party video Polycom System drops the whole conference But no video Audio Dial-in, then Video escalation Case Id No one is currently connected to the conference. Place an Audio call from 8088 (OXE Node 1) to the Third party video System Polycom System conferencing application (dial-in to the btridge). Provides the PIN code through DTMF Once all announcement are finished, 8088 is able escalate to escalate to Video and able to see itself (loopback) or a video prompt. Place an Audio call from a second 8088 (OXE Node 2) to the Third party video System Polycom System conferencing application (dial-in to the btridge). Provides the PIN code through DTMF Once all announcement are finished, second 8088 is able to escalate to Video and see a layout including first 8088 or a video prompt. Not ed ALE Application Partner Program Inter-working report - Edition 1 - page 65/95

66 Several others 8088 (OXE Node 1/2) and Third party video System Endpoints Polycom System(if applicable) joins the Audio/Video conference Video is displayed on first 8088 according to the third party video system dynamic layout algorithm When first 8088 is the active talker, video from first 8088 is seen by other participants according to the third party video system dynamic layout algorithm. Video is displayed on second 8088 according to the third party video system dynamic layout algorithm When second 8088 is the active talker, video from first 8088 is seen by other participants according to the third party video system dynamic layout algorithm Call remains established after waiting 1 minute Place an Audio call from 8088 (OXE Node 1) to the Third party video System Polycom System conferencing application (dial-in to the btridge). Try to escalate to Video as eraly as possible (before providing any code through DTMF, ). Provides the PIN code through DTMF Once all announcement are finished, 8088 is able escalate to see itself (loopback) or a video prompt Audio/Video Dial-out from Third party Video System Case Id Several 8088 and Third party video System Polycom System endpoints (if applicable) are connected to the conference (Audio/Video). User joins the conference through conference URL (or any other means to be called back provided by Third party video System Polycom System) System place an Audio call from conferencing application to the 8088 (OXE Node 1). Answer the call on Once all announcement are finished, audio/video from the other participant can be heared/seen 8088 needs to escalate (see next test) Call remains established after waiting 1 minute Several 8088 and and Third party video System Polycom System endpoints (if applicable) are connected to the conference (Audio/Video). User joins the conference through conference URL (or any other means to be called back provided by Third party video System Polycom System) System place an Audio call from conferencing application to the 8088 (OXE Node 2). Answer the call on Once all announcement are finished, audio/video from the other participant can be heared/seen after 8088 has escalated to Video Call remains established after waiting 1 minute ALE Application Partner Program Inter-working report - Edition 1 - page 66/95

67 Dropped from the conference by Third Party Video System Case Id is connected to the conference (Audio/Video). Third party video System Polycom System drops Third party video System from the scheduled conference is disconnected from the conference and call is released.. ALE Application Partner Program Inter-working report - Edition 1 - page 67/95

68 TUI Menu Case Id DTMF to access Third party video System Polycom System Conference menu Blast Call Case Id A conference is scheduled with pre-defined participants. When the first participants join the conference, all other predefined participants are automatically called by the conference and regular A/V conference takes place. ALE Application Partner Program Inter-working report - Edition 1 - page 68/95

69 8.11 Content Sharing This is to identify behaviour of the system when content sharing capability (if any) is used. Case Id Remark: A scheduled conference is running with one or several Polycom endpoints already connected (A/V). Place an Audio call from a 8088 (OXE Node 1) to the Third party video System Polycom System conferencing application (dial-in to the btridge). Provides the PIN code through DTMF Once all announcement are finished, 8088 is able to escalate to Video and see a layout including others Polycom endpoints. One of the Polycom endpoints start content sharing is able to display the content shared. A scheduled conference is running with one or several Polycom endpoints already connected 5A/V and Content Sharing). Place an Audio call from a 8088 (OXE Node 1) to the Third party video System Polycom System conferencing application (dial-in to the btridge). Provides the PIN code through DTMF Once all announcement are finished, 8088 is able to escalate to Video and see a layout including others is able to display the content shared. - With large external display : Minimum Police size : 10PPI Acceptable Police size : 12PPI - On built-in display : Minimum Police size : 16PPI Acceptable Police size : 18PPI A specific configuration is required on Polycom side. See configuration chapter. A specific configuration is required on Polycom side. See configuration chapter Remote Call Control Case Id - TBD MakeCall (INVITE/Ringing in band/reinvite) Hold/Retrieve Second Call Call Broker Conference All Endpoints except H323. All Endpoints except H323 All Endpoints except H323 All Endpoints except H323 A delay in audio path is noticeable from OXE to Polycom All Endpoints except H323 ALE Application Partner Program Inter-working report - Edition 1 - page 69/95

70 Transfert Release Call All Endpoints except H323 and RealPresence All Endpoints except H OXE Advanced Telephony Services - TBD Case Id Anonymous Call Case Id Place an anonymous call from PSTN extension to Third party video System Polycom System Place an anonymous call from Third party video System Polycom System to PSTN extension Endpoints : OK MCU : OK 8.15 Abandoned Call, Call Routing Error Case Id Place a call from Third party video System Polycom System to a wrong number belonging to OXE network Place a call from Third party video System Polycom System to 8088 (OXE Node 1), Call is abandonned in ringing state Place a call from 8088 (OXE Node 1) to Third party video System Polycom System, Call is abandonned in ringing state Place a call from Third party video System Polycom System to 8088 (OXE Node 1), 8088 is BUSY (no Voic opveflow) Place a call from Third party video System Polycom System to IP Touch (OXE Node 1), 8088 is BUSY (Voic opveflow) Endpoints : OK MCU : OK Endpoints : OK MCU : OK Endpoints : OK MCU : OK Endpoints : OK MCU : OK Endpoints : OK except H.323 MCU : OK (302 ignored, but 1 call blocked in DMA!) ALE Application Partner Program Inter-working report - Edition 1 - page 70/95

71 Place a call from 8088 (OXE Node 1) to Third party video System Polycom System, Third party video System Polycom System is busy Place a call from Third party video System Polycom System to IP Touch (OXE Node 1), 8088 is ringing forever Call automatically stopped by DMA after 25sec 8.16 Tones - TBD Case Id IPv6 - TBD Case Id CAC Case Id OXE H.A. OXE Node 1 is running in geo-redundancy mode. Depending on Third party video System capability, either : - OXE Node name based SIP domain is used to configure relation between Third party video System and OXE Node 1, - Role based OXE virtual IP are used to configure relation between Third party video System and OXE Node 1, Case ALE Application Partner Program Inter-working report - Edition 1 - page 71/95

72 Id OXE Node 1 machine A is role Main. OXE Node 1 machine B is role StdBy Place a call from 8088 OXE Node 1 to Third party video System Polycom System Place a call from Third party video System Polycom System to 8088 OXE Node 1 Change OXE Node 1 machine B role to Main Place a call from 8088 OXE Node 1 to Third party video System Polycom System Place a call from Third party video System Polycom System to 8088 OXE Node 1 Check the downtime between Third party video System Polycom System and OXE Check if Internal DNS resolver is well used by Third party video System Polycom System (TTL to zero) Place a call from Third party video System Polycom System to 8088 OXE Node 1 Reboot OXE Node 1 machine A (Role main given to machine A by priority). Check downtime between Third party video System Polycom System and OXE Less than 1 minutes Keep established communication for 1h Release established communication on Third party video Polycom System side. Release established communication on 8088 OXE Node 1 side. NOK, to check on OXE side Information for End Users This is mainly to identify any existing capability to display a end user would need to use the solution. This is for example a phone number. Case Id Display Extention number that can be dialed on any device (Third party video Polycom System, OXE extensions) ALE Application Partner Program Inter-working report - Edition 1 - page 72/95

73 8.21 Network impairment & Audio/Video quality Video call is established. Several Network impairments profiles are evaluated. Video Quality level selected is Medium Level. Profile A : One way latency : ms, Jitter : 0-50ms, packet lost : % Profile B : One way latency : ms, Jitter : 0-150ms, packet lost : 0 2% Profile C1 : One way latency : ms, Jitter : 0-500ms, packet lost : 0 4% Profile C2 : One way latency : ms, Jitter : 0-500ms, packet lost : 0 10% Case Id An Audio/Video call is established between 8088 (OXE Node 1) and a Third party video System. Network impairement Profile A is used. Check that re-syn/retransmission based on RTCP-FB messages is effective in both direction. An Audio/Video call is established between 8088 (OXE Node 1) and a Third party video System. Network impairement Profile B is used An Audio/Video call is established between 8088 (OXE Node 1) and a Third party video System. Network impairement Profile C1 is used ALE Application Partner Program Inter-working report - Edition 1 - page 73/95

74 9 Appendix A: Polycom Plan for Video Endpoints and Infrastructure cross references. Case # Action ALE Plan ref 1 SIP Server Connectivity 1.1 User Agent (UA) Locates Its SIP Registrar or Proxy Create NAPTR, SRV and A records in DNS pointing to the primary and secondary SIP registrar (and proxy, if applicable). 1.1 to 3.7 NAPTR not 2. Create a SIP account on the registrar for each Polycom supported endpoint with no credentials. 3. Configure each Polycom endpoint with its respective SIP account. TLS not in the scope. Priority on TCP 4. Configure each Polycom endpoint for TLS transport. 5. Configure each Polycom endpoint with the NAPTR record for the SIP registrar (and proxy, if applicable). 6. Verify successful registration of each Polycom endpoint to the primary SIP registrar Repeat test case with the SRV record(s) Repeat test case with the A record(s) Repeat test case with the IP address of the primary SIP registrar (and proxy, if applicable) Repeat test cases for TCP transport Repeat test cases for UDP transport. 1.2 User Agent Failover to Redundant SIP Server Create NAPTR, SRV and A records in DNS pointing to the primary and secondary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint with no credentials. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint for TLS transport. 5. Configure each Polycom endpoint with the NAPTR record for the SIP registrar (and proxy, if applicable). 6. Verify successful registration of each Polycom endpoint to the primary SIP registrar. 7. Disable or disconnect the primary SIP registrar. 8. Verify successful registration of each Polycom endpoint to the secondary SIP registrar. 9. Enable or reconnect the primary SIP registrar. 10. Verify successful registration of each Polycom endpoint to the primary SIP registrar. 11. If using a proxy: Disable or disconnect the primary proxy. 12. If using a proxy: Verify successful registration of each Polycom endpoint to the primary SIP registrar via the secondary proxy. 13. If using a proxy: Enable or reconnect the primary proxy. 14. If using a proxy: Verify successful registration of each Polycom endpoint to the primary SIP registrar via the primary proxy Repeat test case with the SRV record(s) Repeat test case with the A record(s) Repeat test case with the IP address of the primary SIP registrar (and proxy, if applicable) Repeat test cases for TCP transport Repeat test cases for UDP transport. ALE Application Partner Program Inter-working report - Edition 1 - page 74/95

75 Case # Action 1.3 Authentication and Re-Registration Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint with credentials. 3. Configure each Polycom endpoint with its respective SIP account but without credentials. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Verify unsuccessful registration of each endpoint due to authentication failure. 6. Configure each Polycom endpoint with credentials. 7. Verify successful registration of each endpoint. 8. Wait for at least twice the negotiated expiration interval. 9. Verify successful registration of each endpoint. 1.4 Cancellation of Registration Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Verify successful registration of each endpoint. 6. Reboot each Polycom endpoint. 7. Verify successful de-registration of each endpoint before it reboots and re-registration after it reboots. 8. Configure each Polycom endpoint with a different or invalid SIP account. 9. Verify successful de-registration of each endpoint. 2 Point-to-Point Audio and Video Sessions 2.1 Basic Session Establishment Between Two Peers ALE Plan ref ALE Application Partner Program Inter-working report - Edition 1 - page 75/95

76 Case # Action Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Verify successful registration of each endpoint. 6. Initiate a point-to-point call between two endpoints. 7. Verify two-way audio channel using the best available codec (eg. G.719). 8. Verify two-way video channel using the best available codec (eg. H.264), if video is supported by both endpoints. 9. Verify two-way far-end camera control, if supported by both endpoints. 10. Verify content channel from caller to callee, if supported by both endpoints. 11. Verify content channel from callee to caller, if supported by both endpoints. 12. Verify negotiation of LPR via packet captures, if supported by both endpoints. 13. Verify operation of Fast Update via packet captures, if supported by both endpoints. 14. Disconnect call. 15. Repeat steps 6 14 for each pair of endpoints to be tested, in both directions. 2.2 Basic Session Establishment Between Two Peers Security ALE Plan ref 5.1 to 5.3, 5.11, 6.1, 6.5, 6.6, ALE Application Partner Program Inter-working report - Edition 1 - page 76/95

77 Case # Action Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Configure each Polycom endpoint for optional encrypted media. 6. Verify successful registration of each endpoint. 7. Initiate a point-to-point call between two endpoints. 8. Verify encrypted two-way audio channel using the best available codec (eg. G.719). 9. Verify encrypted two-way video channel using the best available codec (eg. H.264), if video is supported by both endpoints. 10. Verify two-way far-end camera control, if supported by both endpoints. 11. Verify encrypted content channel from caller to callee, if supported by both endpoints. 12. Verify encrypted content channel from callee to caller, if supported by both endpoints. 13. Verify negotiation of LPR via packet captures, if supported by both endpoints. 14. Verify operation of Fast Update via packet captures, if supported by both endpoints. 15. Disconnect call. 16. Repeat steps 7 15 for each pair of endpoints to be tested, in both directions Configure each Polycom endpoint for required encrypted media. 2. Repeat test case Unsuccessful Session - No Answer Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Verify successful registration of each endpoint. 6. Initiate a point-to-point call from one endpoint to another, but do not answer. 7. Abandon the call on the calling endpoint. 8. Verify successful tear-down of the call. 9. Repeat steps 7 9 for each pair of endpoints to be tested, in both directions. 2.4 Unsuccessful Session - Busy ALE Plan ref Not Supported ALE Application Partner Program Inter-working report - Edition 1 - page 77/95

78 Case # Action Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Verify successful registration of each endpoint. 6. Initiate a point-to-point call from one endpoint to an endpoint unable to accept more calls. 7. Abandon the call on the calling endpoint. 8. Verify successful tear-down of the call. 9. Repeat steps 7 9 for each pair of endpoints to be tested, in both directions. 2.5 Unsuccessful Session - No Response from User Agent Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Verify successful registration of each endpoint. 6. Initiate a point-to-point call from one endpoint to a SIP account with no UA registered. 7. Verify timely failure of the call and the caller s return to idle state. 8. Repeat steps 6 7 for each endpoint to be tested, in both directions. 2.6 Endpoint Response to Telephony Scenarios Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom endpoint. 3. Configure each Polycom endpoint with its respective SIP account. 4. Configure each Polycom endpoint with the A record for the SIP registrar (and proxy, if applicable). 5. Verify successful registration of each endpoint. 6. Initiate a point-to-point call from one endpoint to a standard SIP telephony device. 7. Verify Hold and Resume functionality initiated by the callee. 8. Verify Hold and Resume functionality initiated by the caller, if supported. 9. Verify Mute and Unmute functionality initiated by the callee. 10. Verify Mute and Unmute functionality initiated by the caller, if supported. 11. Verify Attended and Unattended Transfer functionality initiated by the callee. 12. Verify Attended and Unattended Transfer functionality initiated by the caller, if supported. 13. Repeat steps 6 12 for each endpoint to be tested. 3 Endpoint-to-Bridge Audio & Video Sessions 3.1 Basic Session Establishment Between Endpoint & RMX ALE Plan ref , 7.3, 7.5, 7.6, 7.7, 7.8 ALE Application Partner Program Inter-working report - Edition 1 - page 78/95

79 Case # Action Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom element. 3. Configure each Polycom element with its respective SIP account. 4. Configure each Polycom element with the A record for the SIP registrar (and proxy, if applicable). 5. Create a meeting room on the RMX. 6. Verify successful registration of each element. 7. Initiate calls from each endpoint to the meeting room on the RMX. 8. Verify two-way audio channel using the best available codec on each endpoint (eg. G.719). 9. Verify two-way video channel using the best available codec on each endpoint (eg. H.264), if video is supported on the endpoint. 10. Verify content channel to and from each endpoint, if supported. 11. Verify negotiation of LPR on each endpoint via packet captures, if supported. 12. Verify operation of Fast Update of each endpoint via packet captures, if supported. 13. Verify two-way Far End Camera Control on each endpoint, if supported. 14. Disconnect calls. 3.2 Basic Session Establishment Between Endpoint & RMX - Security Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom element. 3. Configure each Polycom element with its respective SIP account. 4. Configure each Polycom element with the A record for the SIP registrar (and proxy, if applicable). 5. Create a meeting room on the RMX. 6. Configure each Polycom element for optional encrypted media. 7. Verify successful registration of each element. 8. Initiate calls from all endpoints to a meeting room on the RMX. 9. Verify encrypted two-way audio channel using the best available codec on each endpoint (eg. G.719). 10. Verify encrypted two-way video channel using the best available codec on each endpoint (eg. H.264), if video is supported on the endpoint. 11. Verify encrypted content channel to and from each endpoint, if supported. 12. Verify negotiation of LPR on each endpoint via packet captures, if supported. 13. Verify operation of Fast Update of each endpoint via packet captures, if supported. 14. Verify two-way Far End Camera Control on each endpoint, if supported. 15. Disconnect all calls. 16. Configure each Polycom element for required encrypted media. 17. Repeat steps DTMF Monitoring/Control Between Endpoint & RMX ALE Plan ref 10.1, 10.2, 10.3, 10.5, 10.6 Not Supported ALE Application Partner Program Inter-working report - Edition 1 - page 79/95

80 Case # Action Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 2. Create a SIP account on the registrar for each Polycom element. Configure each Polycom element with its respective SIP account. 3. Configure each Polycom element with the A record for the SIP registrar (and proxy, if applicable). 4. Create a meeting room on the RMX. 5. Verify successful registration of each element. 6. Initiate a call from an endpoint to an entry queue on the RMX. 7. Verify that DTMF can be used to select a meeting room. 8. Disconnect call. 9. Repeat steps 7 9 for each endpoint to be tested. 4 Basic Session Establishment Between Endpoint & RMX thru DMA Use DMA as SIP Registrar / Proxy and configure DMA with RMX 2. Create an A record in DNS pointing to the primary SIP registrar (and proxy, if applicable). 3. Create a SIP account on the registrar for each Polycom element. 4. Configure each Polycom element with its respective SIP account. 5. Configure each Polycom element with the A record for the SIP registrar (and proxy, if applicable). 6. Create a meeting room on the DMA. 7. Verify successful registration of each element. 8. Initiate calls from each endpoint to the meeting room on the DMA. 9. Verify two-way audio channel using the best available codec on each endpoint (eg. G.719). 10. Verify two-way video channel using the best available codec on each endpoint (eg. H.264), if video is supported on the endpoint. 11. Verify content channel to and from each endpoint, if supported. 12. Verify negotiation of LPR on each endpoint via packet captures, if supported. 13. Verify operation of Fast Update of each endpoint via packet captures, if supported. 14. Verify two-way Far End Camera Control on each endpoint, if supported. 15. Disconnect calls. 4.1 Basic Session Establishment Between Endpoint & RMX thru DMA - Security ALE Plan ref , 10.2, 10.3, 10.5, 10.6 Not Supported ALE Application Partner Program Inter-working report - Edition 1 - page 80/95

81 Case # Action Use DMA as SIP Registrar / Proxy and configure DMA with RMX 2. Create an A record in DNS pointing to the primary SIP registrar 3. Create a SIP account on the registrar for each Polycom element. 4. Configure each Polycom element with its respective SIP account. 5. Configure each Polycom element with the A record for the SIP registrar. 6. Create a meeting room on the DMA. 7. Configure each Polycom element for optional encrypted media. 8. Verify successful registration of each element. 9. Initiate calls from all endpoints to a meeting room on the DMA. 10. Verify encrypted two-way audio channel using the best available codec on each endpoint (eg. G.719). 11. Verify encrypted two-way video channel using the best available codec on each endpoint (eg. H.264), if video is supported on the endpoint. 12. Verify encrypted content channel to and from each endpoint, if supported. 13. Verify negotiation of LPR on each endpoint via packet captures, if supported. 14. Verify operation of Fast Update of each endpoint via packet captures, if supported. 15. Verify two-way Far End Camera Control on each endpoint, if supported. 16. Disconnect all calls. 17. Configure each Polycom element for required encrypted media. 18. Repeat steps Repeat steps 9 16 using a system registered from external coming through the RPAD 4.2 Basic Session Establishment Between Endpoint & RMX thru DMA - DTMF ALE Plan ref ALE Application Partner Program Inter-working report - Edition 1 - page 81/95

82 Case # Action Use DMA as SIP Registrar / Proxy and configure DMA with RMX 2. Create an A record in DNS pointing to the primary SIP registrar 3. Create a SIP account on the registrar for each Polycom element. 4. Configure each Polycom element with its respective SIP account. 5. Configure each Polycom element with the A record for the SIP registrar. 6. Create a meeting room on the DMA which requires a passcode to join 7. Initiate calls from all endpoints to a meeting room on the DMA. 8. Enter pass code using remote 9. Verify encrypted two-way audio channel using the best available codec on each endpoint (eg. G.719). 10. Verify encrypted two-way video channel using the best available codec on each endpoint (eg. H.264), if video is supported on the endpoint. 11. Verify encrypted content channel to and from each endpoint, if supported. 12. Verify negotiation of LPR on each endpoint via packet captures, if supported. 13. Verify operation of Fast Update of each endpoint via packet captures, if supported. 14. Verify two-way Far End Camera Control on each endpoint, if supported. 15. Disconnect all calls. ALE Plan ref Basic Session Establishment Between Endpoints thru DMA - Failover ALE Application Partner Program Inter-working report - Edition 1 - page 82/95

83 Case # Action Use DMA as SIP Registrar / Proxy and configure 3 rd party SIP Registrar as failover 2. Create an A record in DNS pointing to the primary SIP registrar 3. Create a SIP account on the registrar for each Polycom element. 4. Configure each Polycom element with its respective SIP account. 5. Configure each Polycom element with the A record for the SIP registrar. 6. Create a meeting room on the DMA 7. Initiate calls from all endpoints to a meeting room on the DMA. 8. Verify encrypted two-way audio channel using the best available codec on each endpoint (eg. G.719). 9. Verify encrypted two-way video channel using the best available codec on each endpoint (eg. H.264), if video is supported on the endpoint. 10. Verify encrypted content channel to and from each endpoint, if supported. 11. Verify negotiation of LPR on each endpoint via packet captures, if supported. 12. Verify operation of Fast Update of each endpoint via packet captures, if supported. 13. Verify two-way Far End Camera Control on each endpoint, if supported. 14. Unplug Lan cable from DMA 15. Verify that endpoint register with redundant SIP registrar 16. Verify call stays up 17. Disconnect all calls. ALE Plan ref To be added ALE Application Partner Program Inter-working report - Edition 1 - page 83/95

84 10 Appendix B: AAPP member s Application description Documentation for the Polycom products used to build this solution can be found in the following locations: DMA scheduling/dma_7000.html RMX conferencing_platforms/realpresence_collaboration_server_ve.html Platform Director platform_director/realpresence_platform_director.html HDX html Group up500.html Trio /realpresence_trio.html RealPresence Mobile bile/realpresence_mobile.html ALE Application Partner Program Inter-working report - Edition 1 - page 84/95

85 11 Appendix C: Configuration requirements of the AAPP member s application Add an External SIP Peer (Network->External SIP Peers->Add) entry to link Polycom DMA to your OXE : - External SIP Peers : o Name : <a friendly name for this gateway (for example <OXE Node name>)> o Description : <a description for this gateway> o Type : Other o Next hop address : <OXE Node name FQDN> o Destination Network : Keep it empty o Port : 5060 o Transport Type : TCP o Downgrade: check the option accordingly to you need (OXE does not support sips:) o Prefix range : <Fill here your OXE numbering plan prefix, to allow Polycom System to join OXE resources, including any ARS prefix if needed (for example, 21,0>. o Supports SIP OPTION ping : Check this option - Authentication o Add an entry for the OXE, including OXE real username and password configured in OXE SIP (OXE realm configured in OXE SIP Proxy, and username password configured in OXE SIP External Gateway as incoming username and incoming password>) Configure prefix to join conference (Admin -> Conference Manager -> Conference Settings): - Conference Settings: o Dialing Prefix : <Select a prefix within the Polycom System Numbering Plan : for example : 84 in our example> - There are several options for choosing a prefix, check Admin -> Call Server -> Dial Rules and Admin -> Conference Manager -> Conference Settings. It is possible, as illustrated in the example, to have a single prefix for both endpoints and conference room. By Default, Polycom System Dial Rules will first try to resolve registered endpoint, before trying to search for conference matching the dialed destination. For content sharing, conference templates used to configure conference must be changed the following way (Admin->Conference Manager -> Conferrence Termplates) : - Polycom MCU General Settings o Line rate : select 1472 kbps - Polycom MCU Video Quality o Content Protocol : Select H.264 HD o Send content to legacy enspoints : check this option. ALE Application Partner Program Inter-working report - Edition 1 - page 85/95

86 12 Appendix D: Alcatel-Lucent Enterprise Communication Platform: configuration requirements Homogenous Private Dialling Plan ABC-F IP Link SIP Connexion OXE N2 Numbering plan 22xxxx OXE N1 Numbering Plan : 21xxxx Third party Video System Numbering Plan : 84xxxx In this configuration, Polycom System is seen as a node of the OXE Network. From dial plan perspective, this node is then known to host numbers starting by a unique prefix (Routing Number Prefix). This configuration is suitable for interconnecting OXE to Third Party Video System when this last one is able to expose through the SIP Trunk a homogeneous dialling plan. The SIP Trunk can be configured with either a with fixed or dynamic (Registered) contact address. Sip Authentication is mandatory for communication from Polycom System to the OXE. On OXE Node 1, Check Private SIP Transit Mode System/Other System Parameters/SIP Parameters Private SIP transit mode: Choose Proxy or redirect mode for better interworking (but no CAC/Codec Control), or Full Call Handling mode for CAC/Codec control (but some interworking issue). Create a Routing Number Prefix Translator/Prefix Plan/Create Number : <Prefix used to join the Polycom System (84 for example)> Prefix meaning : Routing No. Network number : <An arbitrary network number associated to the Polycom System> Node Number / ABC-F Trunk Group : <Trunk Group number of SIP Trunk resources to be used> Number of digits : <Total Number of digits to collect before trying to connect to the Polycom System (numbering plan size used on third party Video System), for example : 6, for for up to 9999 different rooms/automated attendant)> Create a SIP External Gateway SIP/SIP Ext Gateway/Create SIP External Gateway ID : A free index Gateway Name : <a friendly name for this gateway (for example <Polycom System Name><location>/<model>)> SIP remote domain : <IP address or FQDN of SIP interface of Polycom System (Polycom DMA 7000)> SIP Port number : <Port number of SIP interface of Polycom System (Polycom DMA 7000 select 5060 by default)> Transport Type : <Transport Protocol to use for SIP interface to Polycom System (select TCP by default)> ALE Application Partner Program Inter-working report - Edition 1 - page 86/95

87 Supervision Timer : <Choose a value> (60 for example) Trunk Group Number : <ABCF SIP Trunk identify of SIP Trunk resources to be used.> DNS Type : DNA A SIP DNS1 IP address : <IP address of a DNS able to resolve Polycom System FQND> (if FQDN is used) SIP DNS2 IP address : <IP address of another DNS able to resolve Polycom System FQND> (if FQDN is used) Outgoing realm : < Polycom System authentication realm (check Call Server device authentication settings)> Outgoing username : < Valid username configured to authenticate toward Polycom System side (check Call Server device authentication settings)> Outgoing password : < Valid password configured to authenticate toward Polycom System side (check Call Server device authentication settings)> Incoming username : <Select a username used to identify Polycom System> Incoming Password : <Select a password used to identify Polycom System> authentication method : SIP Digest Payload type for DTMF : 101 Gateway Type : Standard Type Proxy Identification on IP address : True Video Support Profile : Un Restricted Ignore inactive/black hole : False Network Routing Table Translator/Network Routing Table/Review/Modify (select a free table) Protocol Type : ABC_F Numbering Plan Descriptor : <a free NPD> Associated Ext SIP gateway : <the SIP external gateway to is used to join Polycom System (created above)>. To fix calling number displayed on OXE side (incoming call from Polycom System to OXE extension like IP Touch), you need also to create a External Callback Translation Rule as follow: Ext. Callback Translation Rule Translator/External Numbering Plan/Ext. Callback Translation Table/Descend Hierarchy/Create External Callback Table : 0 (accordingly to your trunk configuration) Basic Number : B< Prefix used to join the Polycom System > (B3 in the example above) No. Digits To Be Removed : 1 Digits To Addd : let empty For a better user experience, if DTMF is required to interact with Polycom System, you can enable option DTMF end-to-end signal at SIP trunk level. Such a way, one the call is established between OXE device and the Polycom System, OXE device switch in DTMF transparency mode automatically. A specific configuration for 8088 need to be done for interworking with polycom : From 8770, go to device manahement. For each 8088, - go to Network tab, and set SIP Parameters/Transport mode to TCP. - go to Device tab, and set H264 RTP packetization mode to NALU as illustrated below : ALE Application Partner Program Inter-working report - Edition 1 - page 87/95

88 - Workaround for Crqms , go to Device tab, and set Max frame size to 1300 For OT Conference, Polycom dial-plan must be known by OpenTouch server. This is done through OXE PBX Ranges configuration, as illustrated below: System Services/Topology/OXE CS/OXE CS Network/OXE CS subnetwork/oxe CS/ Selec the main OXE CS, and add a new PBX range covering Polycom system dial-plan (in the example below, this is range ): ALE Application Partner Program Inter-working report - Edition 1 - page 88/95

89 13 Appendix E: AAPP member s escalation process Polycom has Service Programs available that provide Customers with technical telephone support, advance parts replacement, software upgrades & updates, and access to Polycom s enhanced support portal. These programs are available worldwide and are available through Polycom Sales. ALE Application Partner Program Inter-working report - Edition 1 - page 89/95

90 14 Appendix F: AAPP program 14.1 Alcatel-Lucent Application Partner Program (AAPP) The Application Partner Program is designed to support companies that develop communication applications for the enterprise market, based on Alcatel-Lucent Enterprise's product family. The program provides tools and support for developing, verifying and promoting compliant thirdparty applications that complement Alcatel-Lucent Enterprise's product family. ALE International facilitates market access for compliant applications. The Alcatel-Lucent Application Partner Program (AAPP) has two main objectives: Provide easy interfacing for Alcatel-Lucent Enterprise communication products: Alcatel-Lucent Enterprise's communication products for the enterprise market include infrastructure elements, platforms and software suites. To ensure easy integration, the AAPP provides a full array of standards-based application programming interfaces and fully-documented proprietary interfaces. Together, these enable third-party applications to benefit fully from the potential of Alcatel-Lucent Enterprise products. and verify a comprehensive range of third-party applications: to ensure proper inter-working, ALE International tests and verifies selected third-party applications that complement its portfolio. Successful candidates, which are labelled Alcatel-Lucent Enterprise Compliant Application, come from every area of voice and data communications. The Alcatel-Lucent Application Partner Program covers a wide array of third-party applications/products designed for voice-centric and data-centric networks in the enterprise market, including terminals, communication applications, mobility, management, security, etc. ALE Application Partner Program Inter-working report - Edition 1 - page 90/95

91 Web site The Application Partner Portal is a website dedicated to the AAPP program and where the InterWorking Reports can be consulted. Its access is free at Enterprise.Alcatel-Lucent.com You can access the Alcatel-Lucent Enterprise website at this URL: ALE Application Partner Program Inter-working report - Edition 1 - page 91/95

92 15 Appendix G: AAPP Escalation process 15.1 Introduction The purpose of this appendix is to define the escalation process to be applied by the ALE International Business Partners when facing a problem with the solution certified in this document. The principle is that ALE International Technical Support will be subject to the existence of a valid InterWorking Report within the limits defined in the chapter Limits of the Technical support. In case technical support is granted, ALE International and the Application Partner, are engaged as following: (*) The Application Partner Business Partner can be a Third-Party company or the ALE International Business Partner itself ALE Application Partner Program Inter-working report - Edition 1 - page 92/95