SIP Trunk Solution Sipcall Switzerland Configuration Guideline

Transcription

1 Technical Bulletin OmniPCX Enterprise Release 11.2 Configuration Guideline This document details how to set up an IPBX OXE R for enabling a public SIP trunk with SIP Provider Sipcall by Backbone Solutions AG Switzerland Revision History Edition 1: March 28, 2018 creation of the document Legal notice: The information presented is subject to change without notice. ALE International assumes no responsibility for inaccuracies contained herein. Copyright ALE International 2018

2 Table of contents 1 General References Scope & usage of the configuration guide Scope of Alcatel-Lucent Enterprise s support RFCs supported by OmniPCX Enterprise... 3 SIP RFCs:... 3 RTP, T38 & DTMF (used for SIP)... 4 New RFCs in OXE R and R12.x Software/ Hardware components on customer's infrastructure Supported topology: Feature List & Set Compatibility Supported Features & Sets Sipcall Switzerland SIP Trunk Solution Configuration Signaling protocol and number of physical channels Omnipcx Enterprise configuration Trunk Configuration Trunk Group: Trunk groups local parameters Trunk group NDP selector ARS Configuration ARS Prefix NPD ARS Route List ARS Route Time Based Route List Numbering Command Table SIP Gateway and SIP Proxy Configuration SIP Gateway SIP Proxy SIP Registrar SIP External Gateway Configuration SIP External Gateway SIP trunk configuration abstract Troubleshooting information to be sent to TS contact before tests start up Basic pieces of information to be sent for test failure analysis Service Request (SR) writing for effective assignment and analysis Copyright ALE International 2018 page 2/30

3 1 General This document details the process for configuring from scratch a public SIP trunk of the SIP provider Sipcall Switzerland on a system OXE R References Alcatel-Lucent documentation available on the Business Partner Web Site: [1] Alcatel-Lucent OmniPCX Enterprise Communication Server R11.2 Technical Documentation [2] Technical Bulletin TC2005 Certified SIP providers for OpenTouch and/or OmniPCX Enterprise [3] Troubleshooting Guide TG0069 OmniPCX Enterprise Session Initiation Protocol (SIP) [4] Alcatel-Lucent OpenTouch Session Border Controler R2.3 Recommended Security Guidelines Configuration Note 1.2 Scope & usage of the configuration guide This guide is intended for engineers who are familiar with mgr, OmniVista 8770, OpenTouch and with the very basic set up of the IPBX. Therefore, well-known configurations like that for the IP-LAN or for "Traffic Sharing and Barring" are just reminded without any details. 1.3 Scope of Alcatel-Lucent Enterprise s support The support delivered for this SIP Trunk solution is strictly delimited by the approval context and the system configuration detailed in this document. The protocol and the functional aspects of the SIP trunk are in the scope, but not the audio quality of calls for the part incumbent on the SIP provider or on the client's infrastructure. 1.4 RFCs supported by OmniPCX Enterprise SIP RFCs: RFC 2543 (obsolete by RFC 3261,3262, 3263,3264, 3265): SIP: Session Initiation Protocol RFC 2782: A DNS RR for specifying the location of services (DNS SRV) RFC 2822: Internet Message Format RFC 3261: SIP: Session Initiation Protocol RFC 3262: Reliability of Provisional Responses in SIP (PRACK) RFC 3263: SIP: Locating SIP Servers RFC 3264: An Offer / Answer model with SDP RFC 3265: SIP-Specific Event Notification RFC 3311: The SIP UPDATE Method (session timer only) RFC 3323: Privacy Mechanism for the Session Initiation Protocol (SIP) RFC 3324: Short term requirements for network asserted identity RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks RFC 3265: SIP-specific Event Notification RFC 3515: The Session Initiation Protocol (SIP) Refer method Copyright ALE International 2018 page 3/30

4 RFC 3891/3892: The Session Initiation Protocol (SIP) 'Replaces' Header/ Referred-By Mechanism RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to SIP Mapping RFC 3966: The telephone URI for telephone numbers: since R11 only TEL URI is supported RFC 4497: Inter-working between SIP and QSIG RFC 5373: Requesting Answering Modes for the Session Initiation Protocol RFC 4244: An Extension to the Session Initiation Protocol (SIP)for Request History Information RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP) RFC 3428: Session Initiation Protocol (SIP) Extension for Instant Messaging (partial) RFC 3608: Service Route header RFC 3327: Path Header RFC 1321: Authentication for Outgoing calls RFC 2246: The TLS Protocol Version 1.0 RFC 3268: Advanced Encryption Standard (AES) Cipher suites for Transport Layer Security (TLS) RFC 3280/5280: Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile RFC 3842: A message Summary and Message Waiting Indication Event Package RFC 4028: The session timers in the Session Initiation Protocol RFC 3960: Early Media (partial): Gateway model not supported RFC 4568: Session Description Protocol (SDP) Security Descriptions for Media Streams RFC 5806: Diversion Indication in SIP RFC 3725: Invite without SDP (3pcc in SIP) RFC 3966: The tel URI RFC 5009: The P-Early-Media header RTP, T38 & DTMF (used for SIP) RFC 2617: HTTP Authentication: Basic and Digest Access Authentication RFC 2833/4733: DTMF Transparency. RFC 2833 replaced by RFC 4733 RFC 1889/1890: RTP: A transport protocol for Real-Time applications RFC 2198: RTP Payload for Redundant Audio data RFC 3550: RTP: A Transport Protocol for Real-Time application (audio only) RFC 3551: RTP Profile for Audio and Video Conferences with Minimal Control (audio only) RFC 3711: The Secure Real Time. Supported on A-LU IP Phone and Softphone RFC 3362: T38 ITU-T Procedures for real time Group3 Fax Relay / communications over IP RFC 3711: The Secure Real-time Transport Protocol (SRTP) (media integrity) New RFCs in OXE R and R12.x RFC 4904: Representing Trunk Groups in tel/sip Uniform Resource Identifiers (URIs ) RFC 6140: Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP) RFC 7433 A Mechanism for transporting User to User Call Control Information in SIP draft-ietf-cuss-sip-uui-isdn-08 Interworking ISDN Call Control User Information with SIP Copyright ALE International 2018 page 4/30

5 1.5 Software/ Hardware components on customer's infrastructure INFRA COMPONENT MODEL VERSION (min compatible) OXE OmniPCX Enterprise R L k 1.6 Supported topology: Test-Topology: Copyright ALE International 2018 page 5/30

6 1.7 Feature List & Set Compatibility Supported Features & Sets The following tables list the main inter-operation features and the range of sets that are supported with this SIP Trunk solution. For the different items, refer to the indication given in the support column which is marked as "OK" (for full support), or "WR" (support With Restriction), or "NOK"/ NA (for Not OK or Not Applicable), or "NT" (for Not Tested) TEST CASE # FEATURE TEST / CHECK DESCRIPTION Comment Configure OXE in order to have it registering a range of phone numbers on the public network config menus are available // Comment: OXE does not register all the DID numbers it manages. It only registers its main public identity (or installation number which is in general routed to the attendant) Main Number is registered Phone Number Range: (0-9) / FAILED Configure OXE in order to have it registering a range of phone numbers on the public network / (with or without authentication) / set a correct Configure OXE in order to have it registering a range of phone numbers on the public network / (with or without authentication) / set a bad Registration is OK. "Expires" timer is correctly used for refresh. - SIP UTI looks like : sip:+e164_number@my_sip_provider_dom ain - contact header looks like : sip:+e164_installation_number@ip_addres s_of_ippbx Registration is NOT OK. Check OXE retries to register after "retry-after" timer if sent by operator. Registration Timer set to 480 Re- Registration done after 384 sec. No "retry after" sent by SIPCall. OXE tries to register every 30 sec Configure OXE and operator in order to have SIP trunk in authenticated mode towards the operator (DIGEST authenticated on operator side) config is possible SIPCall configured with Digest Authenticatio n DIGEST authenticated on operator side / correct secret outgoing and incoming calls are possible. Check call flows ( RFC 2617 and RFC 1321 are used for outgoing calls) (see "Some Call flows checks" part) Incoming/Ou tgoing calls possible with correct secret Copyright ALE International 2018 page 6/30

7 1.2.3 DIGEST authenticated on operator side / incorrect secret Outgoing call is impossible. Check call flows ( RFC 2617 and RFC 1321 are used for outgoing calls) Outgoing call not possible, SIPCall sends announceme nt "Falsches Voip Passwort" # 2100 OXE device calls to GSM set : outcall phase and com. establishment ring / (remote or local, please tell) ringback, normal audio after answer. Check the display of the calling number. Remote Ringback # 2101 OXE device calls GSM set: end of com. by local party com. and trunk properly released BYE sent by OXE # 2102 OXE device calls GSM set : end of com. by remote party com. and trunk properly released BYE sent by Provider # 2103 OXE device calls to GSM set : outcall phase and call clearing before answer com. and trunk properly released - Check SIP call flow. CANCEL sent by OXE /OK answered by Network./ then 487 sent by OXE / Ack from network. CANCEL sent by OXE / OK answered by Provider 487 sent by Provider / ACK sent by OXE # 2104 OXE device calls to GSM set : outcall to a wrong number Correct display and trunk properly released Voice Message sent by Provider indicating that dialled number is not assigned Call is not released until caller hangs up # 2105 OXE device calls to GSM set : outcall to a busy set Correct display and busy tone. busy tone heard. (check SIP/SDP traces) Provider sends 488 Busy here # 2110 OXE device calls to PSTN set : outcall phase and com. establishment ring / remote ringback, normal audio after answer Copyright ALE International 2018 page 7/30

8 # 2130 OXE device to public number : outcall with CLIR identity of caller not displayed on callee side. Display of caller shows anonymous when called anwers the call # 2210 incoming public call to OXE device : com. Establishment. End call by any party. display CLI info on callee sides, normal conversation / check SIP trace and from parameter. - Check display is consistent with FROM or PAI parameter. - check SDP Offer/Response exchange - Check Ring back tone on public side Test with inernational call: Display is ok, conversation ok, ringback is availlable, Provider offers G711A and G711U, G729. G711A is used. # 2211 incoming public call to Oxe node 1 set: call clearing before answer display CLI info on caller and callee sides, normal conversation / check SIP trace. - Check SIP call flow. CANCEL sent by network /OK answered by OXE./ then 487 sent by network/ Ack from OXE. Cancel sent by network, OK sent by OXE, 487 Request terminated sent by OXE, ACK sent by network # 2220 Incoming call from public network with secret identity to OXE device. Check that identity is not displayed on Callee side in ringing phase, and in connected phase neither. - Check SIP traces : URI in FROM header should look like : "Anonymous" <sip:anonymous@anonymous.invalid>;tag = Check if P-Asserted ID is avalable in messages PAI is not available - not sent by network It is necessary to set Proxy ID on IP Adress = true in external gateway # 2320 Incoming call: immediate forward to VB (voic -box) CLI and audio-msg in VB of the right callee/ no SIP 302 sent to public network. / possible to navigate in the menus with DTMF Calling number correctly announced, no 302 sent Navigation with DTMF ok Copyright ALE International 2018 page 8/30

9 # 2321 * inc. PSTN : immediate forward to other internal user Audio OK / no SIP 302 message sent to public network. Audio ok, no 302 sent Forwarding desination exists in PAI of OK message - not updated on calling side Call is shown as forwarded call on callee side # 2340 incoming Public call to OXE user : immediate forward to public number display # or name of initial callee on caller side, normal audio after answer / SDP transparency (for supported codecs by OXE) for offer/answer /no SIP 302 message sent to network. /tromboning done on OXE side / check header History Info is present with OXE user number Caller's number correctly received on forwarding destination, history header not used, no 302 messages, RTP flow is direct between external users # 2341 incoming Public call to OXE user: fwd on no answer to public number. display # or name of initial callee on caller side, normal audio after answer / SDP transparency (for supported codecs by OXE) for offer/answer /no SIP 302 message sent to network. /tromboning done on OXE side Caller's number correctly received on forwarding destination, no 302 messages, RTP flow is direct between external users Copyright ALE International 2018 page 9/30

10 # 2350 * inc. PSTN : OXE NODE 1 set Do Not Disturb in SIP traces, 480 temporarily unavailable message is sent to network. It is handled by network in a correct manner. Behaviour dependant on configuration if overflow or not # 2351 # 2352 # 2410 * inc. PSTN : OXE NODE 1 non attributed number incoming public call to busy OXE user incoming public call to OXE set : put on-hold and retrieve communication Call is freed by OXE (404) or overflow to operator (according to OXE config) Normal busy tone is heard on caller side. SIP 486 user busy sent to network. It is handled by network in a correct manner. Music on hold is heard on caller side. Check SIP traces. RE-INVITEs - OK sent between OXE and network. off-hold is OK also. # 2430 public call on going with OXE set : transfer to other OXE internal set display (no evolution of display on public side), Ring back tone on public side. After answer normal audio, clean release no display change on public side ring back tone and voice after transfer ok # 2432 public call on going with OXE set : transfer to ext number tromboning is done on OXE side, no display change on initial caller. Display on second public callee is OXE user, even after answer. RTP flow is direct between both external parties. # 2440 # 2442 # 2451 public call ongoing with an OXE set : transfer to other OXE set (wait for second callee answer before transfering) public call ongoing with an OXE set : transfer to public set (wait for second callee answer before transfering) 2 Public sets and 1 OXE set which has generated conference display (no evolution of display on public side). After transfer normal audio, clean release. Final callee has display of external. tromboning is done on OXE side, no display change on initial caller. Display on second public callee is OXE user. displays are globally OK (the name of all participants is not displayed on public side => normal), normal audio. Check the end of conf is handled correctly Copyright ALE International 2018 page 10/30

11 # 2460 outgoing call to an external IVR from OXE user: DTMF to external server. Check that the DTMF code are well accepted by the remote server / RFC2833 is used RFC 2833 is used, Payload type = 101 Digits are correctly sent, sporadically not recognised correctly on B-Party side # 3130 Incoming call from public network when CAC is saturated. Check call is rejected properly / 503 is corrrecly sent and accepted by network 503 Service unavailable is sent. Provider repeats sending new Invite. No special reaction of Provider (e.g. announceme nt) # 5110 Multiple pages (3) Fax transmit to PSTN from OXE fax machine Check fax transmission is done. Fax mode tested : G711 Only All pages sent and correctly received # 5111 Multiple pages (3) Fax receive from public net to OXE FAX machine Check fax transmission is done. Fax mode tested : G711 Only All pages sent and correctly received Copyright ALE International 2018 page 11/30

12 2 Sipcall Switzerland SIP Trunk Solution Configuration 2.1 Signaling protocol and number of physical channels The SIP trunk uses a specific signaling protocol and some physical resources of the IPBX (i.e. DSP channels). Obviously, it is required a board which provides the system with DSP channels (i.e: GD3/INTIP3 board). NGP boards (INTIP3/GD3) are necessary for fax in G711. It is possible to check the number of DSP channels available in the system by using the command compvisu lio. 2.2 Omnipcx Enterprise configuration Trunk Configuration To enable phone calls over the SIP trunk, it s mandatory to have an ISDN trunk group declared with SIP specification. This can be done in mgr: Trunk Groups -> Trunk Group Copyright ALE International 2018 page 12/30

13 Trunk Group: Mgr Trunk Groups lqreview/modify: Trunk Copyright ALE International 2018 page 13/30

14 Trunk groups local parameters Mgr Trunk Groups Trunk Group lqreview/modify: Trunk Copyright ALE International 2018 page 14/30

15 Trunk group NDP selector Mgr Trunk Groups Trunk Group Trunk group NPD selector ARS Configuration To enable voice calls via the ARS system, it s necessary to have ARS Route lists created via the mgr menu Translator -> Automatic Routing Selection. Several ARS route lists have to be managed for international, and national calls ARS Prefix Mgr Translator Prefix Plan Copyright ALE International 2018 page 15/30

16 NPD The NPD 40 will be used for DDI transcoding Incoming calls and CLIP for National Calls linked to ARS-Table. Mgr Translator External Numbering Numbering Plan Description (NPD) NPD 40 Used Entity for CLIP: Copyright ALE International 2018 page 16/30

17 ARS Route List 2 ARS tables have been created for this configuration. NPD identifier is used to build CLIP. Numbering Command Tabl.Id is used to link ARS table with SIP Gateway. Mgr Translator Automatic Route Selection ARS Route List National Calls within Switzerland International Calls Copyright ALE International 2018 page 17/30

18 ARS Route Mgr Translator Automatic Route Selection ARS Route List ARS Route Quality is Speech and Fax Copyright ALE International 2018 page 18/30

19 Quality is Speech and Fax Warning The management of the ARS for Emergency calls (e.g. 112) is not shown here Time Based Route List Mgr Translator Automatic Route Selection ARS Route List Time Based Route List Copyright ALE International 2018 page 19/30

20 Identical for ARS Route list Numbering Command Table Mgr Translator Automatic Route Selection Numbering Command Table Copyright ALE International 2018 page 20/30

21 2.2.3 SIP Gateway and SIP Proxy Configuration SIP Gateway Mgr SIP SIP Gateway SIP Proxy mgr SIP SIP Proxy Copyright ALE International 2018 page 21/30

22 SIP Registrar mgr SIP SIP Registrar SIP External Gateway Configuration SIP External Gateway mgr SIP SIP Ext Gateway Copyright ALE International 2018 page 22/30

23 Copyright ALE International 2018 page 23/30

24 Copyright ALE International 2018 page 24/30

25 2.2.5 SIP trunk configuration abstract The following tables gather the overall system configuration. They only show the values to be modified, that means that the values that are not appearing here will be the default system values. NOTE: NPD here is meant for incoming calls. Outgoing calls will use NPD managed in ARS routes. Several ARS routes have been managed for national, international. System SIP parameters Path: System / Other System Param. / SIP Parameters default value new value (if modified) Packetization times per codec True Via Header_ Inbound Calls Routing False TLS signaling possible False Local resources True Loose Route with RegID True False Reject unidentified proxy calls SRTP offer answer mode False Hotel doorcam application False Transfer : Refer using single step True RE-INVITE delay for hold 3 SIP Bearer Capability Speech Number of SIP trunks (UCaaS) 0 10 Enhanced codec negotiation Local Type Not Available G722 for SIP trunking True sipmotor restart delay 5 Private SIP transit mode Mixed mode SIP registered pseudo reservation False Blind transfer with direct RTP True From Header For Anonymous Calls Anomymous Maximum Trunk Group Overflow 3 SIP video transit mode Not Available Raise SIP Motor Incidents Enhanced Canonical Form SIP UUI Normal Transit Force NCT on Internal Route SIP diversion info for incoming System compression parameters Path: System / Other System Param. / Compression default value new value (if modified) Voice Activity Detect (Comp Bds) False Post Filter on Compressor Boards True Compression Type G 723 G729 Volume on boards with compressors 8 Copyright ALE International 2018 page 25/30

26 VRE on boards with compressors Multi. Algorithms for Compression Voice Activity Detection on G711 G722 data rate G722 Conference With OMS False False False 64 K True Additional System parameters Path: System / Other System Param. / System Parameters default value new value (if modified) Law A-Law Accept Mu and A laws in SIP False DTMF in RFC2833 Only False Path: System/Other System Param./External Signaling Parameters NPD for external forward Calling Name Presentation : False IP parameters Path: IP / IP Parameters default value new value (if modified) Fast Start True Round trip delay request True Jitter buff size(modem/fax transp) 40 G711 VOIP Framing 20 ms G729 VOIP Framing 20 ms G723 VOIP Framing 30 ms Jitter algorithm (voice) 1 Jitter buffer size (voice) 30 DTMF mode 0 CAC with OTMS/OTBE False Path: IP / Fax Parameters T38 only True Local T38 port number RTP port number RTP port number + 3 NAT Support for FAX T38 False Timers used by SIP trunk group Path: External Services / Trunk COS default value new value (if modified) Trunk COS : T2 T0 ABC-F ISDN Trunks Timer T Timer T Timer T Copyright ALE International 2018 page 26/30

27 Timer T Timer T Timer T Timer T Timer T Timer T ABC-F Trunks Timer T Timer T Timer T Timer T Timer T Timer T Timer T Timer T Timer T Timer T Timer T Copyright ALE International 2018 page 27/30

28 3 Troubleshooting 3.1 information to be sent to TS contact before tests start up Before going through the test plan, we strongly recommends to place and inbound and outbound and log the following commands/logs and send the results to SIP Tech Support in order to make sure the initial configuration is correct. a. infocollect b. motortrace for both inbound and outbound calls motortrace 3 traced > /tmpd/tracesip& 3.2 Basic pieces of information to be sent for test failure analysis Call scenario description (calling, called number.) infocollect Traces: - For all cases: Network traces of SIP dialogs (on OXE side in case of NAT translation) OXE SIP motor traces: killall traced motortrace 3 traced -1 /tmpd/sipmotortrace_ s f 99 -d & - Call Handling level 2 traces: Use cases: Signaling problems: Unsuccessful calls, inconsistent SIP messages content, bad codec negotiation, RTP stream setup failure These traces will help for call setup problems deeper analysis of VoIP problems understanding. tuner km tuner ctr trc i Copyright ALE International 2018 page 28/30

29 tuner clear-traces tuner all=off actdbg all=off tuner +at +tr +s +cpu +cpl hybrid=on actdbg sip=on fct=on isdn=on abcf=on rtp=on actdbg cnx=on voip=on mtracer -ag -1 /tmpd/sipchtrace_ f 99 -s d& These are basic traces. Other traces may be requested after first problem description and analysis. 3.3 Service Request (SR) writing for effective assignment and analysis SR creation must follow the rules hereafter: A SR must be opened for any problem. One SR must describe one problem only. SR must contain detailed test case description: I. Test reference in test plan if any, test summary otherwise. II. III. IV. Identification of the elements involved in the test (MCDU, IP address ). Elements role in the test (callee, caller, OXE user, external user called/calling through SIP trunk, automated attendant, SIP gateway, SBC ). Elements important settings in VoIP context (used encoding law, compression algorithm). V. Step by step test description. VI. Detailed test result: Call establishment failure, immediate call cut, half way audio from caller to callee, no audio, bad audio quality, bad display on called set For efficient routing to Technical Support structure dedicated to SIP interoperability tests follow-up: - SR summary must contain the following tag: SBD-<Tested Provider name> - SR status summary/short description must contain: SR followed by: Mr X. - Immediately after SR creation, a mail should be sent to your TS contact with SR identifier and short summary. Copyright ALE International 2018 page 29/30

30 Follow us on Facebook and Twitter Connect with us on Facebook and Twitter for the latest: Software releases Technical communications AAPP InterWorking reports Newsletters and much more! twitter.com/alue_care facebook.com/alecustomercare Submitting a Service Request Please connect to our eservice Request application. Before submitting a Service Request, please be sure: The application has been certified via the AAPP if a third party application is involved. You have read the release notes that list new features, system requirements, restrictions, and more, and are available in the Technical Documentation Library. You have read through the related troubleshooting guides and technical bulletins available in the Technical Documentation Library. You have read through the self-service information on commonly asked support questions and known issues and workarounds available in the Technical Knowledge Center. - END OF DOCUMENT - Copyright ALE International 2018 page 30/30