Avaya Design Team Avaya MPS Design Team 31 st May 2013 Version 1.0. Media Processing Server (MPS) to Session Manager (SM) Interoperability

Transcription

1 Media Processing Server (MPS) to Session Manager (SM) Interoperability Avaya Design Team Avaya MPS Design Team 31 st May 2013 Version 1.0 Avaya 2013 Last Saved: 31 May 2013 Issue: 1.0 Page 1 of Avaya Inc. - All Rights Reserved

2 Revision History Date Version Status Description 05/31/ Released Original Avaya 2013 Last Saved: 31-May-13 Issue: 1.0 Status: Draft Page 2 of 15

3 Content 1. PURPOSE INTRODUCTION EQUIPMENT AND SOFTWARE VALIDATED PREREQUISITE KNOWLEDGE CONFIGURE AVAYA AURA SESSION MANAGER ADD A LOCATION FOR MPS SPECIFY THE LISTEN PORT FOR SESSION MANAGER TO COMMUNICATION WITH MPS DEFINE MPS ENTITY DEFINE ENTITY LINK BETWEEN SESSION MANAGER AND MPS DEFINE ROUTING POLICY AND DIAL PATTERN FOR MPS CONFIGURE AVAYA MEDIA PROCESSING SERVER MPS COMPONENT (MPS AP) CCSS SIP /OPT/CCSS/ETC/SIPDEFAULT.PROPERTIES /OPT/CCSS/ETC/SIP.CONF CALL SCENARIOS VERIFIED MPS INTER-OPERATES WITH SESSION MANAGER AND CM-FS APPLICATION SEQUENCE MPS INTER-OPERATES WITH SESSION MANAGER AND WITHOUT CM-FS APPLICATION SEQUENCE Issue: 1.0 Page 3 of 15

4 1. Purpose This interoperability guide describes configuration of a network that uses SIP Trunks between Avaya Aura Session Manager Release 6.3 and Avaya Media Processing Server 4.1 using Common Channel Signalling Server (CCSS) configured for SIP protocol. Avaya Aura Session Manager provides SIP proxy/routing functionality, routing SIP sessions across an IP network with centralized routing policies and registrations for SIP endpoints. Avaya Media Processing Server 4.1 is a DTMF and speech enabled interactive voice response system that supports VoiceXML2.x compliant DTMF / speech applications and MPS Developer DTMF / Speech applications. MPS CCSS SIP provides SIP telephony interface to MPS Application Processor. This guide provides information for the setup, configuration, and verification of the call flows tested on this solution. Issue: 1.0 Page 4 of 15

5 2. Introduction This interoperability guide describes configuration of a network that uses SIP Trunks between Avaya Aura Session Manager Release 6.3 and Avaya Media Processing Server 4.1 using CCSS configured for SIP protocol. In the sample configuration calls from different types of stations registered to Avaya Session Manager are routed to by Session Manager to the Media Processing Server platform where the caller can interact with self service applications. Validation was performed for users registered to Session Manager with and without using CM-FS as a sequenced application for the user to simulate both user calls and calls from another SIP entity such as an AudioCodes Mediant 3000 Gateway to PSTN for example. The following MPS features and functionality have been validated in the sample configuration: Inbound calls o Accept inbound calls o Reject inbound calls o Redirect inbound calls Transfer calls: o Bridge transfer o Consultation transfer (SIP Attended transfer) o Blind transfer (Unattended transfer) o Transfer calls with attached data Hold and Retrieve calls Fast start and Slow start calls Supported DTMF types o RFC2833 (Recommended) o Inband o SIP INFO* Supported codec types o G711 (Recommended for speech recognition) o G729 UPDATE message SIP Outbound calls to various clients configured with CM/SM including H323, SIP client and other SIP entity *NOTE: Having a known issue with SIP INFO for DTMF: wi CCSS SIP should be enhanced to handle application/dtmf INFO messages as DTMFs in DSSI mode MPS Developer applications have been validated with both DSSI (TMS integrated) and PROXY (ASE) modes for all functionalities described above. VoiceXML applications have been validate with DSSI mode for inbound calls only. Issue: 1.0 Page 5 of 15

6 TDM SIP SIP Issue: 1.0 Page 6 of 15

7 3. Equipment and Software Validated The following equipment and software were used for the sample configuration. Component Version Avaya Aura Session Manager (SM) Avaya Aura System Manager (SMGR) Avaya Communication Manager Feature Server CM6.2 SP5 Avaya Media Processing Server 500 (MPS) MPS 4.1 (Beta Load) Avaya Common Channel Signalling Server (CCSS) MPS 4.1 (Beta Load) AudioCodes Mediant A Avaya One-X Communicator SP SJ phone Note: Although not validated MPS version 3.5 is expected to have similar results as the MPS 4.1. Issue: 1.0 Page 7 of 15

8 4. Prerequisite Knowledge These application notes assume the reader has a certain level of knowledge of the installation and administration of Avaya Media Processing Server and Avaya Aura Session Manager. Furthermore, these application notes were written from the perspective that the installation of the above-mentioned components has already been completed. Issue: 1.0 Page 8 of 15

9 5. Configure Avaya Aura Session Manager This section describes the steps needed to configure Session Manager to route calls to and from MPS. These instructions assume that the Session Manager is installed and configured and SIP users added. 5.1 Add a Location for MPS 5.2 Specify the Listen port for Session Manager to communication with MPS Make sure that the Session Manger Asset is configured Issue: 1.0 Page 9 of 15

10 5.3 Define MPS Entity Define MPS Link under SIP Entities and commit 5.4 Define Entity Link between Session Manager and MPS Go back to MPS entity link created in the previous step and go to Entity link section to create an entity link between Session Manager and MPS Note: MPS Entity must be configured for UDP and Trusted. Issue: 1.0 Page 10 of 15

11 5.5 Define Routing Policy and Dial Pattern for MPS Configure routing policy and dial pattern as per customer configuration requirement Issue: 1.0 Page 11 of 15

12 6. Configure Avaya Media Processing Server 6.1 MPS Component (MPS AP) No specific MPS component configuration changes are required for Session Manager. 6.2 CCSS SIP The MPS and CCSS SIP server can be configured in SIP proxy or DSSI mode /opt/ccss/etc/sipdefault.properties Add the following entries to this file: sip.longformat.enable=true udp.buffer.size=5120 mps.api.header.all=true sip.notify.autorsp=true mg.codec.selection=first mps.contaddr.max.count= /opt/ccss/etc/sip.conf The following specific items need to be configured: Client access needs to be set to UDP on port 5060 CLIENT_ACCESS = {! ! NAME TYPE LOCAL TIMER! PORT PROFILE! CLIENT UDP 5060 SIP_TMR1 } Set up a network host for the session manager security asset using the domain configured in Session Manager. NETWORK_HOSTS = { SM1 UDP Billerica-sm.avaya.com SIP_TMR1 false } No registration to Session Manager is required as MPS is configured as trusted entity. Issue: 1.0 Page 12 of 15

13 7. Call Scenarios Verified MPS Developer applications have been validated with both DSSI (TMS integrated) and PROXY (ASE) modes for all scenarios described below. VoiceXML applications have been validate with DSSI mode for inbound calls only. 7.1 MPS inter-operates with Session Manager and CM-FS Application Sequence Test case Description / Notes Pass/Fail Accept inbound call from PSTN/H323/SIP client then blind/consultation transfer to H323/SIP/Hunt-Group The call comes from PSTN/H323/SIP through SM and is routed to MPS SIP. MPS SIP answers this call then transfers it to H323/SIP client or CM Hunt-Group. MPS SIP can use blind Reject/Redirect inbound call from H323/SIP Client Transfer and send data to H323/SIP Client Outbound call to SIP/H323/Hunt-Group Client Hold and retrieved the call with SIP/H323/Hunt-Group Client Receive Fast/Slow start inbound call from SIP/H323 client Make Fast/Slow start outbound call to SIP/H323 client Make Voice Bridge between PSTN/SIP/H323 clients Send/receive Inband DTMF to/from SIP/H323 client Send/receive RFC2833 DTMF to/from SIP/H323 client Send/receive SIP-INFO DTMF to/from SIP/H323 client Speak/recognize the voice with PCMA/PCMU/G729 coder to/from SIP/H323 client or consultation to transfer call. When the call comes from H323/SIP client to MPS SIP. MPS SIP can reject or redirect this call MPS SIP can blind/consultation transfer a call to H323/SIP client with data attached by using Refer-To with UUI. UUI info which is sent from MPS SIP to H323/SIP clients must be encoded as HEX and have prefixed: C8. EX: Hello is encoded as 68656c6c6f, so the UUI string is C868656c6c6f MPS SIP makes an outbound call to SIP/H323 client or CM Hunt-Group MPS SIP hold and retrieve the call with SIP/H323 client or CM Hunt- Group MPS SIP can answer an inbound call with Fast/Slow start model MPS SIP can make an outbound call with Fast/Slow start model Client A make calls to MPS SIP. MPS answer this call and make a new call to Client B. Then MPS SIP can bridge the voice of A and B together. While interacting with client, MPS SIP can send and receive DTMF. While interacting with client, MPS SIP can send and receive DTMF. While interacting with client, MPS SIP can send and receive DTMF. While interacting with client, MPS SIP can: - Speak with PCMA/PCMU/G729 coder. - Recognize with G711 coder FAILED Issue: 1.0 Page 13 of 15

14 Send/receive SIP re-invite message to/from H323/SIP client Send out SIP UPDATE message to SIP/H323 client Accept hold/retrieve call from SIP/H323 client Receive SIP OPTION heartbeat from SM Call transfer between SIP client and H323 client Transfer call from SIP/H323 client to Application Transfer call to CM/SM client - No answer Transfer call to CM/SM client - Not found When the call is established between MPS SIP and client. MPS SIP can accept or send SIP re-invite message When the call is established between MPS SIP and client. MPS SIP can send SIP UPDATE message When the call is established between MPS SIP and client. Client can hold and retrieve the call After configuring MPS SIP as an SIP endpoint of SM. SM will send ping message (SIP OPTION) to monitor status of MPS SIP. MPS SIP can receive and respond SIP 200 OK message for this OPTION automatically. When the call is established between MPS SIP and client. Client can transfer the call to another client. When the call is established between client1 and client2. Client1 can transfer the call to MPS SIP. Call comes from PSTN/H323/SIP through SM and is routed to MPS SIP. MPS SIP answers this call then transferring it to H323/SIP client. MPS SIP can use blind or consultation to transfer call to another client but that client is no answer. Call comes from PSTN/H323/SIP through SM and is routed to MPS SIP. MPS SIP answers this call then transferring it to H323/SIP client MPS SIP can use blind or consultation to transfer call to another client but that client is not valid 7.2 MPS inter-operates with Session Manager and without CM-FS Application Sequence Test case Description / Notes Pass/Fail Accept/reject/redirect inbound call from When the call comes from other SIP other SIP endpoint endpoint to MPS SIP. MPS SIP can accept, reject or redirect this call. Receive Fast/Slow start inbound call from MPS SIP can answer an inbound call other SIP endpoint with Fast/Slow start model Make Fast/Slow start outbound call to other MPS SIP can make an outbound call SIP endpoint with Fast/Slow start model Send/receive Inband DTMF to/from other While interacting with client, MPS SIP SIP endpoint Send/receive RFC2833 DTMF to/from other SIP endpoint can send and receive DTMF. While interacting with client, MPS SIP can send and receive DTMF. Issue: 1.0 Page 14 of 15

15 Send/receive SIP INFO DTMF to/from other SIP endpoint Speak/recognize the voice with PCMA/PCMU/G729 coder to/from other SIP endpoint Send/receive SIP re-invite message to/from other SIP endpoint Receive SIP OPTION heartbeat from SM While interacting with client, MPS SIP can send and receive DTMF. While interacting with other SIP endpoint, MPS SIP can: - Speak with PCMA/PCMU/G729 coder. - Recognize with G711 coder When the call is established between MPS SIP and other SIP endpoint. MPS SIP can accept or send SIP re- INVITE message After configuring MPS SIP as an SIP endpoint of SM. SM will send ping message (SIP OPTION) to monitor status of MPS SIP. MPS SIP can receive and respond SIP 200 OK message for this OPTION automatically. FAILED Issue: 1.0 Page 15 of 15