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@pbx1.nucleus.ngn.vodafone.de: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@nbgsx001.ngn.vodafone.de;user=phone From: sip:CALLING_NUMBER@ims.vodafone.de;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@ims.vodafone.de;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@ims.vodafone.de;user=phone;tag=SDvao3a01-e398456e To: sip:MY_EXTERNAL_NUMBER@nbgsx001.ngn.vodafone.de;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 [cid:NUCLEUS-ohne-Untertext-schwarz-160px_b2f8f234-adb9-4bf7-9d12-97f87894f25e11.png] Marc Tschech m.tschech@nucleus-gmbh.com Phone: +49 (211) 415559-0 | Fax: +49 (211) 415559-29 www.nucleusultrasonics.com<http://www.nucleusultrasonics.com> [cid:pic_c3124c36-ec5f-4ec5-8b32-9fac6c7fb0b3111.png] NUCLEUS GmbH * Tichauer Weg 32 * 40231 Düsseldorf * Germany Geschäftsführer: M. Tschech Amtsgericht Düsseldorf HRB 5540 This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. [cid:NUCLEUS-ohne-Untertext-schwarz-160px_b2f8f234-adb9-4bf7-9d12-97f87894f25e11.png] Marc Tschech m.tschech@nucleus-gmbh.com Phone: +49 (211) 415559-0 | Fax: +49 (211) 415559-29 www.nucleusultrasonics.com<http://www.nucleusultrasonics.com> [cid:pic_c3124c36-ec5f-4ec5-8b32-9fac6c7fb0b3111.png] NUCLEUS GmbH * Tichauer Weg 32 * 40231 Düsseldorf * Germany Geschäftsführer: M. Tschech Amtsgericht Düsseldorf HRB 5540 This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet.

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@pbx1.nucleus.ngn.vodafone.de: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@nbgsx001.ngn.vodafone.de;user=phone > From: sip:CALLING_NUMBER@ims.vodafone.de;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@ims.vodafone.de;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@ims.vodafone.de;user=phone;tag=SDvao3a01-e398456e > To: sip:MY_EXTERNAL_NUMBER@nbgsx001.ngn.vodafone.de;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 >

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. I tested with the service number of Vodafone (498001721234) and got exactly this SDP in 200 OK or 183 Session Progress: v=0 o=- 626692240 761358328 IN IP4 x.y.z.q s=IMSS c=IN IP4 a.b.c.d t=0 0 m=audio 11258 RTP/SAVP 8 101 a=rtpmap:101 telephone-event/8000 a=sendrecv a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:.... => Seems to be a "feature" of Vodafone. But this call seemed to work for me (I could hear the IVR system). The original call, which was reproducibly broken, worked until the MOH was interrupted by a human being (I didn't wait for a Vodafone agent coming in). Thanks Michael > > > <- History Entry 569 Received from 88.79.204.9:5060 at 1563208936 -> > INVITE sip:MY_EXTERNAL_NUMBER@pbx1.nucleus.ngn.vodafone.de: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@nbgsx001.ngn.vodafone.de;user=phone > From: sip:CALLING_NUMBER@ims.vodafone.de;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@ims.vodafone.de;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