Hi, I've noticed a one-way audio problem with one of the SIP providers. It's definitely a media issue, not a network issue. I make a call, another party answers and hears me, but I can't her another party. When he or she starts speaking, the output on my side says "strm0x98fd74 Bad RTP pt 8 (expecting 117)" I've noticed that there are different codec priorities in 183 and in 200 replies from that provider. When it replies with 183 with iLBC as the most preferred codec, pjsua-lib sais pjsua_media.c Media updates, stream #0: iLBC (sendrecv) Later, when provider replies with 200 and PCMA as the most preferred codec, there are no "Media updates" in the pjsua-lib log output and there is "SDP negotiation done, message body is ignored" instead. And error messages start appearing when another party speaks. Is such behaviour of that SIP provider appropriate? Should pjsua-lib adapt to the codec changes between 1xx and 2xx? Alexei 17:20:38.866 pjsua_core.c TX 1318 bytes Request msg INVITE/ cseq=31721 (tdta0x85fa00) to UDP 217.73.112.14:5060: INVITE sip:012345678901 at sip.pctel.ru SIP/2.0 Via: SIP/2.0/UDP 91.78.58.52:5060;rport;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb Max-Forwards: 70 From: sip:johnsmith@xxxxxxxxxxxx;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3 To: sip:012345678901 at sip.pctel.ru Contact: <sip:johnsmith at 91.78.58.52:5060> Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e CSeq: 31721 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: PJSUA v1.0.2/i386-apple-darwin9.6.0 Authorization: Digest username="johnsmith", realm="sip.pctel.ru", nonce="49edc952a4cf810a00e6bae7252c2707601bf829", uri="sip:012345678901 at sip.pctel.ru ", response="c8a54e3a5f7d1d4a9e15549f46f527e3", cnonce="hsrQ4p0kagPMvuwq6OMEMpnGuEIMX9re", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 456 v=0 o=- 3449308838 3449308838 IN IP4 91.78.58.52 s=pjmedia c=IN IP4 91.78.58.52 t=0 0 a=X-nat:8 m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101 a=rtcp:4001 IN IP4 91.78.58.52 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 17:20:38.866 pjsua_app.c Call 0 state changed to CALLING 17:20:38.906 pjsua_core.c RX 309 bytes Response msg 100/INVITE/ cseq=31721 (rdata0x82b064) from UDP 217.73.112.14:5060: SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 91.78.58.52 :5060;rport=5060;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb From: sip:johnsmith@xxxxxxxxxxxx;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3 To: sip:012345678901 at sip.pctel.ru Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e CSeq: 31721 INVITE Content-Length: 0 --end msg-- 17:20:39.105 pjsua_core.c RX 793 bytes Response msg 183/INVITE/ cseq=31721 (rdata0x82b064) from UDP 217.73.112.14:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 91.78.58.52 :5060;rport=5060;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb From: sip:johnsmith@xxxxxxxxxxxx;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3 To: sip:012345678901 at sip.pctel.ru;tag=as4dac19c9 Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e CSeq: 31721 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:012345678901 at 217.73.112.9:5060> Content-Type: application/sdp Content-Length: 291 v=0 o=root 30663 30663 IN IP4 217.73.112.9 s=session c=IN IP4 217.73.112.9 t=0 0 m=audio 18218 RTP/AVP 117 8 0 3 101 a=rtpmap:117 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --end msg-- 17:20:39.105 pjsua_app.c Call 0 state changed to EARLY (183 Session Progress) 17:20:39.128 strm0x865b74 VAD temporarily disabled 17:20:39.129 strm0x865b74 Encoder stream started 17:20:39.129 strm0x865b74 Decoder stream started 17:20:39.129 pjsua_media.c Media updates, stream #0: iLBC (sendrecv) 17:20:39.129 conference.c Port 3 (sip:012345678901 at sip.pctel.ru) transmitting to port 0 (Built-in Microphone) 17:20:39.129 conference.c Port 0 (Built-in Microphone) transmitting to port 3 (sip:012345678901 at sip.pctel.ru) 17:20:39.129 pjsua_app.c Media for call 0 is active 17:20:39.131 Master/sound Underflow, buf_cnt=0, will generate 1 frame 17:20:39.778 strm0x865b74 VAD re-enabled 17:20:49.469 pjsua_core.c RX 820 bytes Response msg 200/INVITE/ cseq=31721 (rdata0x82b064) from UDP 217.73.112.14:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.78.58.52 :5060;rport=5060;branch=z9hG4bKPjH4fxoJD4c4kpDnZbuWrNO2THj01FD-Jb Record-Route: <sip:217.73.112.14;lr=on> From: sip:johnsmith@xxxxxxxxxxxx;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3 To: sip:012345678901 at sip.pctel.ru;tag=as4dac19c9 Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e CSeq: 31721 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:012345678901 at 217.73.112.9:5060> Content-Type: application/sdp Content-Length: 291 v=0 o=root 30663 30664 IN IP4 217.73.112.9 s=session c=IN IP4 217.73.112.9 t=0 0 m=audio 18218 RTP/AVP 8 117 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:117 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --end msg-- 17:20:49.470 pjsua_app.c Call 0 state changed to CONNECTING 17:20:49.470 inv0x85bc64 SDP negotiation done, message body is ignored 17:20:49.470 pjsua_core.c TX 388 bytes Request msg ACK/cseq=31721 (tdta0x86c200) to UDP 217.73.112.14:5060: ACK sip:012345678901 at 217.73.112.9:5060 SIP/2.0 Via: SIP/2.0/UDP 91.78.58.52:5060;rport;branch=z9hG4bKPjnUzGc4KND2SpWxshSKHX8DMbQ1numik6 Max-Forwards: 70 From: sip:johnsmith@xxxxxxxxxxxx;tag=GX09pOKuM86R75zTInh0NIq.M06tHfG3 To: sip:012345678901 at sip.pctel.ru;tag=as4dac19c9 Call-ID: tXloaqtu3fCEUw9fnE0EHxAJpWHGoh9e CSeq: 31721 ACK Route: <sip:217.73.112.14;lr> Content-Length: 0 --end msg-- 17:20:49.470 pjsua_app.c Call 0 state changed to CONFIRMED 17:20:49.616 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.626 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.641 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.668 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.684 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.704 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.724 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.741 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.761 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.784 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.804 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.824 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.842 strm0x865b74 Bad RTP pt 8 (expecting 117) 17:20:49.861 strm0x865b74 Bad RTP pt 8 (expecting 117)
