Hi, The 200/OK response says the call will use GSM (RTP PT 3), but we're receiving PCMU (RTP PT 0). Either callee or B2BUA in the middle is not sending us correct RTP packets. Best regards, ?Benny On Wed, Sep 22, 2010 at 7:31 PM, Chiang Kang Tan <chiang.tan at malaysiaparadise.co.uk> wrote: > Hi all, > > I'm seeing this "Bad RTP pt" error which is mentioned in the FAQ as well. > > My remote-endpoint is a Softphone called X-Lite 4, which is the latest > version I can find. > > Below are INVITE, 200 OK response, and the error message. > > Any idea why I keep seeing the error messages, even when I've set the codec > priority of PCMU/8000 to the highest? (In fact I don't quite understand why > GSM/8000 is being used instead as well) > > Thank you. > > Chiang > > 13:20:52.006?? pjsua_core.c? TX 1053 bytes Request msg INVITE/cseq=5750 > (tdta0x92d31f8) to UDP 192.168.1.99:5060: > INVITE sip:2003 at asterix.domain SIP/2.0 > Via: SIP/2.0/UDP > 192.168.1.19:5060;rport;branch=z9hG4bKPjd71d7bad-af3a-4b55-8f23-9a77f69d813a > Max-Forwards: 70 > From: sip:wibble@asterix.domain;tag=8900c1d2-1351-4efe-b104-2462f2a321bf > To: sip:2003 at asterix.domain > Contact: <sip:wibble at 192.168.1.19:5060> > Call-ID: eb2a1fa4-fc3d-4718-89b5-fdc9eb76fec0 > CSeq: 5750 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Session-Expires: 1800 > Min-SE: 90 > Content-Type: application/sdp > Content-Length:?? 459 > > v=0 > o=- 3494146852 3494146852 IN IP4 192.168.1.19 > s=pjmedia > c=IN IP4 192.168.1.19 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 0 103 102 109 3 104 8 9 101 > a=rtcp:4001 IN IP4 192.168.1.19 > a=rtpmap:0 PCMU/8000 > a=rtpmap:103 speex/16000 > a=rtpmap:102 speex/8000 > a=rtpmap:109 iLBC/8000 > a=fmtp:109 mode=30 > a=rtpmap:3 GSM/8000 > a=rtpmap:104 speex/32000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 13:20:54.180?? pjsua_core.c? TX 819 bytes Response msg 200/INVITE/cseq=102 > (tdta0x92e4d50) to UDP 192.168.1.99:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.1.99:5060;rport=5060;received=192.168.1.99;branch=z9hG4bK6056328e > Call-ID: eb2a1fa4-fc3d-4718-89b5-fdc9eb76fec0 > From: <sip:2003@asterix.domain>;tag=as6233b853 > To: <sip:wibble at asterix.domain>;tag=8900c1d2-1351-4efe-b104-2462f2a321bf > CSeq: 102 INVITE > Session-Expires: 1800;refresher=uas > Contact: <sip:wibble at 192.168.1.19:5060> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Supported: replaces, 100rel, timer, norefersub > Content-Type: application/sdp > Content-Length:?? 250 > > v=0 > o=- 3494146852 3494146853 IN IP4 192.168.1.19 > s=pjmedia > c=IN IP4 192.168.1.19 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 3 101 > a=rtcp:4001 IN IP4 192.168.1.19 > a=rtpmap:3 GSM/8000 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 13:20:54.254? strm0x92dc5f4? Bad RTP pt 0 (expecting 3) > > > _______________________________________________ > 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 > >
