[SR-Users] Issues with Yealink SDP and sdp_with_transport()

[Daniel-Constantin Mierla]

Hello,

can you provide the sdp bodies for both Grandstream (that matched) and
Yealink (that didn't match). We have to compare how the SAVP is
advertised and how the function is making the check.

Cheers,
Daniel

On 08/07/15 16:45, Alberto Sagredo wrote:
>
> Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is
> send SAVP or not to divert to and rtp proxy or rtpengine, as you know
> rtpproxy supports recording and rtpengine does not yet. 
>
> So when using  if(sdp_with_transport("RTP/SAVP")) with Grandstream
> Phones all worked fine, but when configuring Optional or Compulsory
> SRTP in Yealink it seems to do not detect 
>
>
> i have seen that crypto lines are not in the final SDP but do not know
> if thats the reason
>
> Did you have a similar issue with Yealink?
>
> If i could get traces in anyway to help let me know.
>
>
> BR
>
>
> Alberto
>
>
> INVITE sip:212 at 10.0.1.34:15060 <http://sip:212@10.0.1.34:15060> SIP/2.0.
>
> Record-Route: <sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes>.
>
> Record-Route:
> <sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes>.
>
> Via: SIP/2.0/UDP
> x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1.
>
> Via: SIP/2.0/TLS
> 10.0.1.111:11891;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360.
>
> From: "214" <sip:214 at 1x.x.x.x:8001>;tag=1072578853.
>
> To: <sip:212 at x.x.x.x:8001>.
>
> Call-ID: 0_1310998066 at 10.0.1.111 <mailto:0_1310998066 at 10.0.1.111>.
>
> CSeq: 2 INVITE.
>
> Contact: <sip:214 at 83.x.x.x:11891;transport=TLS>.
>
> Content-Type: application/sdp.
>
> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY,
> REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
>
> Max-Forwards: 69.
>
> User-Agent: Yealink SIP-T21P_E2 52.80.0.3.
>
> Allow-Events: talk,hold,conference,refer,check-sync.
>
> Content-Length:   553.
>
> .
>
> v=0.
>
> o=- 20143 20143 IN IP4 x.x.x.x.
>
> s=SDP data.
>
> c=IN IP4 x.x.x.x
>
> t=0 0.
>
> m=audio 8530 RTP/AVP 0 8 18 101.
>
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm.
>
> a=crypto:2 AES_CM_128_HMAC_SHA1_32
> inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3.
>
> a=crypto:3 F8_128_HMAC_SHA1_80
> inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw.
>
> a=rtpmap:0 PCMU/8000.
>
> a=rtpmap:8 PCMA/8000.
>
> a=rtpmap:18 G729/8000.
>
> a=fmtp:18 annexb=no.
>
> a=rtpmap:101 telephone-event/8000.
>
> a=fmtp:101 0-15.
>
> a=ptime:20.
>
> a=sendrecv.
>
> a=nortpproxy:yes
>
>
>
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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com

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