pjsua is known to not support asymmetric codecs - it expects to receive the same codec as it sends. regards klaus Alexei Kuznetsov schrieb: > 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) > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
