Re: missing rtpmap returns BAD Request

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



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@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


_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@xxxxxxxxxxxxxxx
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org



[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux