Hello Marc, I'm seeing here exactly the same (Telekom AllIP customer) - but the impact seems to be different: I can hear no audio (I'm not sure if the peer had one). Which version of pjsip are you using? I'm using 2.9 w/ asterisk 16.5. Do you probably know of a phone number which shows the problem and can be used to test against? If it's an IVR system that shouldn't be a problem. The number I have unfortunately is not an IVR system ... . Thanks Michael On 25.07.19 at 11:07 Marc Tschech wrote: > Hey guys, > > I have the following problem: > I have a business sip-trunk from Vodafone Germany. On a few incoming calls I see that there is a mistake in the incoming INVITE: > > As you can see there is m=audio 55004 RTP/AVP 96 9 8 101 102 > So an rtpmap for 96, 101 and 102 is required. > Unfortunately the 96 mapping is missing. I would expect to ignore the 96 codec and work with one of the other but instead PJSIP is returning BAD REQUEST and the call drops. > > > <- History Entry 569 Received from 88.79.204.9:5060 at 1563208936 -> > INVITE sip:MY_EXTERNAL_NUMBER@xxxxxxxxxxxxxxxxxxxxxxxxxxxx:5060 SIP/2.0 > Via: SIP/2.0/UDP 88.79.204.9:5060;received=88.79.204.9;branch=z9hG4bKrkmtgr009837vfa01600.1;origin=172.19.116.80 > To: sip:MY_EXTERNAL_NUMBER@xxxxxxxxxxxxxxxxxxxxxxxx;user=phone > From: sip:CALLING_NUMBER@xxxxxxxxxxxxxxx;user=phone;tag=SDvao3a01-e398456e > Call-ID: SDvao3a01-7b27dc6a53070acda0d03a904428df03-ct4u830000 > CSeq: 1 INVITE > Max-Forwards: 60 > Contact: sip:CALLING_NUMBER@88.79.204.9:5060;transport=udp > Date: Mon, 15 Jul 2019 18:42:16 GMT > Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE > Supported: resource-priority > P-Asserted-Identity: sip:CALLING_NUMBER@xxxxxxxxxxxxxxx;user=phone > P-Asserted-Identity: tel:CALLING_NUMBER > Accept: application/sdp > P-Early-Media: supported > Content-Type: application/sdp > Content-Length: 289 > Content-Type: application/sdp > Content-Length: 289 > > v=0 > o=- 0 0 IN IP4 88.79.204.9 > s=IMSS > c=IN IP4 88.79.204.9 > t=0 0 > m=audio 55004 RTP/AVP 96 9 8 101 102 > a=rtpmap:101 telephone-event/8000 > a=rtpmap:102 telephone-event/16000 > a=ptime:20 > a=maxptime:30 > a=fmtp:96 mode-set=0,1,2; mode-change-period=2; mode-change-neighbor=1; max-red=0 > > > > <- History Entry 570 Sent to 88.79.204.9:5060 at 1563208936 -> > SIP/2.0 400 Bad Request > Via: SIP/2.0/UDP 88.79.204.9:5060;rport=5060;received=88.79.204.9;branch=z9hG4bKrkmtgr009837vfa01600.1;origin=172.19.116.80 > Call-ID: SDvao3a01-7b27dc6a53070acda0d03a904428df03-ct4u830000 > From: sip:CALLING_NUMBER@xxxxxxxxxxxxxxx;user=phone;tag=SDvao3a01-e398456e > To: sip:MY_EXTERNAL_NUMBER@xxxxxxxxxxxxxxxxxxxxxxxx;user=phone;tag=z9hG4bKrkmtgr009837vfa01600.1 > CSeq: 1 INVITE > Warning: 399 SIP "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" > Server: FPBX-14.0.13.4(15.4.0) > Content-Length: 0 > _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@xxxxxxxxxxxxxxx http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
