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

Alberto Sagredo alberto.sagredo at avanzada7.com
Tue Jul 14 14:40:59 CEST 2015


Hi Daniel

Here its Yealink one (Optional SRTP)

If you need anything more let me know

INVITE sip:1 at 192.168.0.181:5080 SIP/2.0.

Record-Route: <sip:x.x.x.x 8002;r2=on;lr=on;ftag=4139505128;nat=yes>.

Record-Route:
<sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=4139505128;nat=yes>.

Via: SIP/2.0/UDP
x.x.x.x:8002;branch=z9hG4bK4928.64d27b89d3de27a54610b8ebb2aa9f43.0;i=2b2.

Via: SIP/2.0/TLS 10.0.1.111:11880
;received=x.x.x.x;rport=11880;branch=z9hG4bK1819432518.

From: "214" <sip:214 at x.x.x.x:8001>;tag=4139505128.

To: <sip:1 at x.x.x.x:8001>.

Call-ID: 0_3807548115 at 10.0.1.111.

CSeq: 2 INVITE.

Contact: <sip:214 at 80.x.x.x:11880;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:   549.

.

v=0.

o=- 20005 20005 IN IP4 192.168.0.178.

s=SDP data.

c=IN IP4 192.168.0.178.

t=0 0.

m=audio 8546 RTP/AVP 0 8 18 101.

a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:MjI1MDczY2JjYTM4MjM0MyBlMmIyZGI2YmUyZWI1.

a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:N2EwZjhkMjAxMjlkMmFjMjcyY2E5NDczODM3Yjdh.

a=crypto:3 F8_128_HMAC_SHA1_80
inline:IDQ2YTBiYzQ2MDA1Y2ZhYWNkNTZmNmQ5NWY4Yjcw.

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.


And a GS one (Optional SRTP)


U 192.168.0.170:8002 -> 192.168.0.181:5080

INVITE sip:2 at 192.168.0.181:5080 SIP/2.0.

Record-Route: <sip:x.x.x.x:8002;lr=on;ftag=429447500;nat=yes>.

Via:
SIP/2.0/UDP x.x.x.x:8002;branch=z9hG4bKc08.2bf157be2b7c1a44c1128d55db60357c.0.

Via:
SIP/2.0/UDP x.x.x.x:46597;received=x.x.x.x;branch=z9hG4bK1529661043;rport=46597.

From: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=429447500.

To: <sip:2 at x.x.x.x:8002>.

Call-ID: 2055647556-46597-5 at IA.CG.BIE.BCH.

CSeq: 40 INVITE.

Contact: "Anonymous" <sip:212 at x.x.x.x:46597>.

X-Grandstream-PBX: true.

Max-Forwards: 69.

User-Agent: Grandstream GXP2140 1.0.4.23.

Privacy: id.

P-Preferred-Identity: <sip:212 at x.x.x.x:8002>.

Supported: replaces, path, timer.

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.

Content-Type: application/sdp.

Accept: application/sdp, application/dtmf-relay.

Content-Length:   753.

.

v=0.

o=212 8000 8000 IN IP4 x.x.x.x

s=SIP Call.

c=IN IP4  x.x.x.x.

t=0 0.

m=audio 32584 RTP/AVP 0 8 18 9 2 101.

a=sendrecv.

a=rtpmap:0 PCMU/8000.

a=ptime:20.

a=rtpmap:8 PCMA/8000.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=no.

a=rtpmap:9 G722/8000.

a=rtpmap:2 G726-32/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-15.

m=audio 32584 RTP/SAVP 0 8 18 9 2 101.

a=sendrecv.

a=rtpmap:0 PCMU/8000.

a=ptime:20.

a=rtpmap:8 PCMA/8000.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=no.

a=rtpmap:9 G722/8000.

a=rtpmap:2 G726-32/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-15.

a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:/cVB/SqgmIibo+CJTVZvnDNOf9dNxFFaQc70pqbm.

a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:76OrMKDV0Dhda9w+9SmUZMbHskWe/wnwWUq+TfFk.

2015-07-13 9:19 GMT+02:00 Daniel-Constantin Mierla <miconda at gmail.com>:

>  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 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.
>
> 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
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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