Hi, Our client using PJSIP (v2.0.1) is experiencing lost audio on resuming a held incoming call and at first glance it seems to be due to SDP records sent with in-dialog SIP INVITEs. The client initially receives an SDP offer in the incoming SIP INVITE from a third party SIP client: <snip> m=audio 35198 RTP/AVP 98 99 3 8 0 101 13 a=rtpmap:98 SPEEX/16000 a=rtpmap:99 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <snip> The PJSIP client responds with the negotiated SDP in the 200OK <snip> m=audio 40000 RTP/AVP 98 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:98 speex/16000 <snip> Placing the call on hold some time later with pjsua_call_set_hold however generates and sends an in-dialog INVITE including an SDP record: <snip> m=audio 40000 RTP/AVP 103 102 3 8 0 101 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <snip> Changing the mapping of dynamic payload types for speex/16000 and speex/8000 from the initial SDP offer seems contrary to section 8.3.2 of RFC3264 - An Offer/Answer Model with the Session Description Protocol (SDP) (https://tools.ietf.org/html/rfc3264#section-8.3.2) Subsequently resuming this call with pjsua_call_reinvite also issues an INVITE with an SDP record similar to that generated by pjsua_call_set_hold. The result is log messages from PJSIP "strm0x871ff844 Bad RTP pt 98 (expecting 103)" and audio is heard in only one direction. Am I missing something or should I suspect PJSIP? Thanks for any help. Regards, Chris Gibson. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131113/346da534/attachment-0001.html>
