Hello, ? I have a question regarding codec negotiation between a Mac OS PjSip-based client and Counterpath's X-lite. I've tried the scenario with both the 1.10 and 2.0 versions of PjSip. I'm using version 4.1 of the X-lite client. The scenario is as follows: ? 1. Initiate call from PjSip client to X-Lite client 2. PjSip offers list of supported codecs 3. X-Lite responds with acceptable codecs (note multiple) 4. PjSip attempts to update the session to use only one codec 5. ?X-Lite continues to respond with a list 6. Steps 4-5 are repeated until session is disconnected. ? Note that if either X-Lite or Pjsip are configured to use only one codec, this does not happen and everything works as expected. Below is a PjSip log snippet showing the offer/response loop.. ?Am I doing something wrong here? Is there a bug in X-Lite or PjSip? Any suggestions would be very much appreciated. ? Thanks! Vlad ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:848] RX 1041 bytes Response msg 200/INVITE/cseq=8107 (rdata0x104010a28) from UDP 192.168.8.226:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 135.225.30.6:55270;rport;branch=z9hG4bKPjCPCkvdkoygfMaqRq9wtg0f.GmmaQ3XAi;received=135.225.30.6 From: sip:9001@192.168.8.226;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU To: <sip:9002 at 192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 CSeq: 8107 INVITE Contact: <sip:192.168.8.226:5060> X-Genesys-CallUUID: 7N9O2JBD1579J0DFBGG65HLGCC0000AU Allow: INVITE, ACK, PRACK, CANCEL, BYE, UPDATE User-Agent: X-Lite 4 release 4.1 stamp 63214 Session-Expires: 1800;refresher=uas Min-SE: 90 Supported: uui,timer Content-Type: application/sdp Content-Length: 356 ? v=0 o=- 1316129799 1 IN IP4 192.168.5.49 s=CounterPath c=IN IP4 192.168.5.49 t=0 0 a=ice-ufrag:dc4360 a=ice-pwd:ed1a2843eaf9680fc8dbb7da226f99e1 m=audio 51096 RTP/AVP 0 8 96 a=sendrecv a=candidate:1 1 UDP 659136 192.168.5.49 51096 typ host a=candidate:1 2 UDP 659134 192.168.5.49 51097 typ host a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 ? --end msg-- ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:849] VAD temporarily disabled ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:849] Encoder stream started ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:849] Decoder stream started ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:849] Media updates, stream #0: PCMU (sendrecv) ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [INFO] [13:38:41:849] Got answer with multiple codecs, scheduling updating media session to use only one codec.. ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:849] Port 1 (sip:9002 at 192.168.8.226:5060;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV) transmitting to port 0 (Built-in Microphone) ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:849] Port 0 (Built-in Microphone) transmitting to port 1 (sip:9002 at 192.168.8.226:5060;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV) ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:850] TX 412 bytes Request msg ACK/cseq=8107 (tdta0x101857e00) to UDP 192.168.8.226:5060: ACK sip:192.168.8.226:5060 SIP/2.0 Via: SIP/2.0/UDP 135.225.30.6:55270;rport;branch=z9hG4bKPjAqPsiCTlI27uOfh5cxwdvn8jm-5Cm5o2 Max-Forwards: 70 From: sip:9001@192.168.8.226;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU To: <sip:9002 at 192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 CSeq: 8107 ACK Content-Length: ?0 ? ? --end msg-- ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [INFO] [13:38:41:850] Updating media session to use only one codec.. ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:850] TX 792 bytes Request msg UPDATE/cseq=8108 (tdta0x101878c00) to UDP 192.168.8.226:5060: UPDATE sip:192.168.8.226:5060 SIP/2.0 Via: SIP/2.0/UDP 135.225.30.6:55270;rport;branch=z9hG4bKPjI6.sAbIP2xc.6.U9VupEjlVm4f3hs8Yo Max-Forwards: 70 From: sip:9001@192.168.8.226;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU To: <sip:9002 at 192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 Contact: <sip:9001 at 135.225.30.6:55270;ob> Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 CSeq: 8108 UPDATE Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: ? 248 ? v=0 o=- 3525626319 3525626320 IN IP4 135.225.30.6 s=pjmedia c=IN IP4 135.225.30.6 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 0 96 a=rtcp:4003 IN IP4 135.225.30.6 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 ? --end msg-- ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:894] Underflow, buf_cnt=0, will generate 1 frame ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:993] RX 1073 bytes Request msg INVITE/cseq=1 (rdata0x101875228) from UDP 192.168.8.226:5060: INVITE sip:9001 at 135.225.30.6:55270;ob SIP/2.0 From: <sip:9002@192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 To: sip:9001 at 192.168.8.226;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 CSeq: 1 INVITE Content-Length: 367 Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.8.226:5060;branch=z9hG4bK58150E47-ED75-47E9-AAE8-B4E674BDA2BB-2053 Contact: <sip:192.168.8.226:5060> Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO, UPDATE, MESSAGE, NOTIFY, OPTIONS X-Genesys-CallUUID: 7N9O2JBD1579J0DFBGG65HLGCC0000AU Max-Forwards: 70 Session-Expires: 1800;refresher=uac Min-SE: 90 Supported: 100rel,timer ? v=0 o=- 1316129799 2 IN IP4 192.168.5.49 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.5.49 t=0 0 a=ice-ufrag:dc4360 a=ice-pwd:ed1a2843eaf9680fc8dbb7da226f99e1 m=audio 51096 RTP/AVP 0 8 96 a=sendrecv a=candidate:1 1 UDP 659136 192.168.5.49 51096 typ host a=candidate:1 2 UDP 659134 192.168.5.49 51097 typ host a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 ? --end msg-- ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:41:993] TX 413 bytes Response msg 491/INVITE/cseq=1 (tdta0x10400fc00) to UDP 192.168.8.226:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.8.226:5060;received=192.168.8.226;branch=z9hG4bK58150E47-ED75-47E9-AAE8-B4E674BDA2BB-2053 Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 From: <sip:9002@192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 To: <sip:9001 at 192.168.8.226>;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU CSeq: 1 INVITE Content-Length: ?0 ? ? --end msg-- ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:42:054] RX 401 bytes Request msg ACK/cseq=1 (rdata0x104019e28) from UDP 192.168.8.226:5060: ACK sip:9001 at 135.225.30.6:55270;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.8.226:5060;branch=z9hG4bK58150E47-ED75-47E9-AAE8-B4E674BDA2BB-2053 From: <sip:9002@192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 To: <sip:9001 at 192.168.8.226>;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 CSeq: 1 ACK Content-Length: 0 ? ? --end msg-- ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:42:251] Bad RTP pt 126 (expecting 0) ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:42:251] Bad RTP pt 126 (expecting 0) ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:42:255] Bad RTP pt 126 (expecting 0) ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:42:350] TX 792 bytes Request msg UPDATE/cseq=8108 (tdta0x101878c00) to UDP 192.168.8.226:5060: UPDATE sip:192.168.8.226:5060 SIP/2.0 Via: SIP/2.0/UDP 135.225.30.6:55270;rport;branch=z9hG4bKPjI6.sAbIP2xc.6.U9VupEjlVm4f3hs8Yo Max-Forwards: 70 From: sip:9001@192.168.8.226;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU To: <sip:9002 at 192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 Contact: <sip:9001 at 135.225.30.6:55270;ob> Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 CSeq: 8108 UPDATE Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: ? 248 ? v=0 o=- 3525626319 3525626320 IN IP4 135.225.30.6 s=pjmedia c=IN IP4 135.225.30.6 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 0 96 a=rtcp:4003 IN IP4 135.225.30.6 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 ? --end msg-- ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:42:474] VAD re-enabled ? [Sample.GSSipEndpoint.PjSipProvider.PjSip] [DEBUG] [13:38:43:157] RX 1073 bytes Request msg INVITE/cseq=2 (rdata0x10087e628) from UDP 192.168.8.226:5060: INVITE sip:9001 at 135.225.30.6:55270;ob SIP/2.0 From: <sip:9002@192.168.8.226;genesysid=7N9O2JBD1579J0DFBGG65HLGCC0000AV>;tag=FFEE5D7C-19CE-423A-9DF4-12D5826507DF-832 To: sip:9001 at 192.168.8.226;tag=ldEe8hFdWOgPp1o8yA7L67rNtmAYBdCU Call-ID: T-IpkxdpkLV0MRuJSB1UewoYVKAfCXj1 CSeq: 2 INVITE Content-Length: 367 Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.8.226:5060;branch=z9hG4bK58150E47-ED75-47E9-AAE8-B4E674BDA2BB-2054 Contact: <sip:192.168.8.226:5060> Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO, UPDATE, MESSAGE, NOTIFY, OPTIONS X-Genesys-CallUUID: 7N9O2JBD1579J0DFBGG65HLGCC0000AU Max-Forwards: 70 Session-Expires: 1800;refresher=uac Min-SE: 90 Supported: 100rel,timer ? v=0 o=- 1316129799 2 IN IP4 192.168.5.49 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.5.49 t=0 0 a=ice-ufrag:dc4360 a=ice-pwd:ed1a2843eaf9680fc8dbb7da226f99e1 m=audio 51096 RTP/AVP 0 8 96 a=sendrecv a=candidate:1 1 UDP 659136 192.168.5.49 51096 typ host a=candidate:1 2 UDP 659134 192.168.5.49 51097 typ host a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 ? --end msg-- ? -------------- next part -------------- An HTML attachment was scrubbed... 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