Hi Daniel
Here its Yealink one (Optional SRTP)
If you need anything more let me know
INVITE sip:1@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@x.x.x.x:8001>;tag=4139505128.
To: <sip:1@x.x.x.x:8001>.
Call-ID: 0_3807548115@10.0.1.111.
CSeq: 2 INVITE.
Contact: <sip:214@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@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@anonymous.invalid>;tag=429447500.
To: <sip:2@x.x.x.x:8002>.
Call-ID: 2055647556-4659...@ia.cg.bie.bch.
CSeq: 40 INVITE.
Contact: "Anonymous" <sip:212@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@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 <mico...@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@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@1x.x.x.x:8001>;tag=1072578853.
>
> To: <sip:212@x.x.x.x:8001>.
>
> Call-ID: 0_1310998066@10.0.1.111.
>
> CSeq: 2 INVITE.
>
> Contact: <sip:214@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-us...@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@lists.sip-router.org
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>
>
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