<div dir="ltr">Hi Daniel<div><br></div><div>Here its Yealink one (Optional SRTP)</div><div><br></div><div>If you need anything more let me know</div><div>
<p class=""><span class="">INVITE <a href="http://sip:1@192.168.0.181:5080">sip:1@192.168.0.181:5080</a> SIP/2.0.</span></p>
<p class=""><span class="">Record-Route: <sip:x.x.x.x 8002;r2=on;lr=on;ftag=4139505128;nat=yes>.</span></p>
<p class=""><span class="">Record-Route: <sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=4139505128;nat=yes>.</span></p>
<p class=""><span class="">Via: SIP/2.0/UDP x.x.x.x:8002;branch=z9hG4bK4928.64d27b89d3de27a54610b8ebb2aa9f43.0;i=2b2.</span></p>
<p class=""><span class="">Via: SIP/2.0/TLS 10.0.1.111:11880;received=x.x.x.x;rport=11880;branch=z9hG4bK1819432518.</span></p>
<p class=""><span class="">From: "214" <sip:214@x.x.x.x:8001>;tag=4139505128.</span></p>
<p class=""><span class="">To: <sip:1@x.x.x.x:8001>.</span></p>
<p class=""><span class="">Call-ID: <a href="mailto:0_3807548115@10.0.1.111">0_3807548115@10.0.1.111</a>.</span></p>
<p class=""><span class="">CSeq: 2 INVITE.</span></p>
<p class=""><span class="">Contact: <sip:214@80.x.x.x:11880;transport=TLS>.</span></p>
<p class=""><span class="">Content-Type: application/sdp.</span></p>
<p class=""><span class="">Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.</span></p>
<p class=""><span class="">Max-Forwards: 69.</span></p>
<p class=""><span class="">User-Agent: Yealink SIP-T21P_E2 52.80.0.3.</span></p>
<p class=""><span class="">Allow-Events: talk,hold,conference,refer,check-sync.</span></p>
<p class=""><span class="">Content-Length: 549.</span></p>
<p class=""><span class="">.</span></p>
<p class=""><span class="">v=0.</span></p>
<p class=""><span class="">o=- 20005 20005 IN IP4 192.168.0.178.</span></p>
<p class=""><span class="">s=SDP data.</span></p>
<p class=""><span class="">c=IN IP4 192.168.0.178.</span></p>
<p class=""><span class="">t=0 0.</span></p>
<p class=""><span class="">m=audio 8546 RTP/AVP 0 8 18 101.</span></p>
<p class=""><span class="">a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MjI1MDczY2JjYTM4MjM0MyBlMmIyZGI2YmUyZWI1.</span></p>
<p class=""><span class="">a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:N2EwZjhkMjAxMjlkMmFjMjcyY2E5NDczODM3Yjdh.</span></p>
<p class=""><span class="">a=crypto:3 F8_128_HMAC_SHA1_80 inline:IDQ2YTBiYzQ2MDA1Y2ZhYWNkNTZmNmQ5NWY4Yjcw.</span></p>
<p class=""><span class="">a=rtpmap:0 PCMU/8000.</span></p>
<p class=""><span class="">a=rtpmap:8 PCMA/8000.</span></p>
<p class=""><span class="">a=rtpmap:18 G729/8000.</span></p>
<p class=""><span class="">a=fmtp:18 annexb=no.</span></p>
<p class=""><span class="">a=rtpmap:101 telephone-event/8000.</span></p>
<p class=""><span class="">a=fmtp:101 0-15.</span></p>
<p class=""><span class="">a=ptime:20.</span></p>
<p class=""><span class="">a=sendrecv.</span></p>
<p class=""><span class="">a=nortpproxy:yes.</span></p><p class=""><span class=""><br></span></p><p class=""><span class="">And a GS one </span>(Optional SRTP)</p><p class=""><span class=""><br></span></p><p class=""><span class="">U <a href="http://192.168.0.170:8002">192.168.0.170:8002</a> -> <a href="http://192.168.0.181:5080">192.168.0.181:5080</a></span></p><p class=""><span class="">INVITE <a href="http://sip:2@192.168.0.181:5080">sip:2@192.168.0.181:5080</a> SIP/2.0.</span></p><p class=""><span class="">Record-Route: <sip:x.x.x.x:8002;lr=on;ftag=429447500;nat=yes>.</span></p><p class=""><span class="">Via: SIP/2.0/UDP x.x.x.x:8002;branch=z9hG4bKc08.2bf157be2b7c1a44c1128d55db60357c.0.</span></p><p class=""><span class="">Via: SIP/2.0/UDP x.x.x.x:46597;received=x.x.x.x;branch=z9hG4bK1529661043;rport=46597.</span></p><p class=""><span class="">From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=429447500.</span></p><p class=""><span class="">To: <sip:2@x.x.x.x:8002>.</span></p><p class=""><span class="">Call-ID: 2055647556-46597-5@IA.CG.BIE.BCH.</span></p><p class=""><span class="">CSeq: 40 INVITE.</span></p><p class=""><span class="">Contact: "Anonymous" <sip:212@x.x.x.x:46597>.</span></p><p class=""><span class="">X-Grandstream-PBX: true.</span></p><p class=""><span class="">Max-Forwards: 69.</span></p><p class=""><span class="">User-Agent: Grandstream GXP2140 1.0.4.23.</span></p><p class=""><span class="">Privacy: id.</span></p><p class=""><span class="">P-Preferred-Identity: <sip:212@x.x.x.x:8002>.</span></p><p class=""><span class="">Supported: replaces, path, timer.</span></p><p class=""><span class="">Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE.</span></p><p class=""><span class="">Content-Type: application/sdp.</span></p><p class=""><span class="">Accept: application/sdp, application/dtmf-relay.</span></p><p class=""><span class="">Content-Length: 753.</span></p><p class=""><span class="">.</span></p><p class=""><span class="">v=0.</span></p><p class=""><span class="">o=212 8000 8000 IN IP4 x.x.x.x</span></p><p class=""><span class="">s=SIP Call.</span></p><p class=""><span class="">c=IN IP4 x.x.x.x.</span></p><p class=""><span class="">t=0 0.</span></p><p class=""><span class="">m=audio 32584 RTP/AVP 0 8 18 9 2 101.</span></p><p class=""><span class="">a=sendrecv.</span></p><p class=""><span class="">a=rtpmap:0 PCMU/8000.</span></p><p class=""><span class="">a=ptime:20.</span></p><p class=""><span class="">a=rtpmap:8 PCMA/8000.</span></p><p class=""><span class="">a=rtpmap:18 G729/8000.</span></p><p class=""><span class="">a=fmtp:18 annexb=no.</span></p><p class=""><span class="">a=rtpmap:9 G722/8000.</span></p><p class=""><span class="">a=rtpmap:2 G726-32/8000.</span></p><p class=""><span class="">a=rtpmap:101 telephone-event/8000.</span></p><p class=""><span class="">a=fmtp:101 0-15.</span></p><p class=""><span class="">m=audio 32584 RTP/SAVP 0 8 18 9 2 101.</span></p><p class=""><span class="">a=sendrecv.</span></p><p class=""><span class="">a=rtpmap:0 PCMU/8000.</span></p><p class=""><span class="">a=ptime:20.</span></p><p class=""><span class="">a=rtpmap:8 PCMA/8000.</span></p><p class=""><span class="">a=rtpmap:18 G729/8000.</span></p><p class=""><span class="">a=fmtp:18 annexb=no.</span></p><p class=""><span class="">a=rtpmap:9 G722/8000.</span></p><p class=""><span class="">a=rtpmap:2 G726-32/8000.</span></p><p class=""><span class="">a=rtpmap:101 telephone-event/8000.</span></p><p class=""><span class="">a=fmtp:101 0-15.</span></p><p class=""><span class="">a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/cVB/SqgmIibo+CJTVZvnDNOf9dNxFFaQc70pqbm.</span></p><p class="">
</p><p class=""><span class="">a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:76OrMKDV0Dhda9w+9SmUZMbHskWe/wnwWUq+TfFk.</span></p></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-07-13 9:19 GMT+02:00 Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Hello,<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<div>On 08/07/15 16:45, Alberto Sagredo
wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">
<p><span>Im using </span>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. </p>
<p>So when using <span> </span>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 </p>
<p><br>
i have seen that crypto lines are not in the final SDP but do
not know if thats the reason</p>
<p>Did you have a similar issue with Yealink?<br>
<br>
If i could get traces in anyway to help let me know.</p>
<p><br>
</p>
<p>BR</p>
<p><br>
Alberto</p>
<p><span><br>
</span></p>
<p><span>INVITE <a href="http://sip:212@10.0.1.34:15060" target="_blank">sip:212@10.0.1.34:15060</a>
SIP/2.0.</span></p>
<p><span>Record-Route:
<a><sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes></a>.</span></p>
<p><span>Record-Route:
<a><sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes></a>.</span></p>
<p><span>Via: SIP/2.0/UDP
x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1.</span></p>
<p><span>Via: SIP/2.0/TLS
10.0.1.111:11891;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360.</span></p>
<p><span>From: "214"
<a><sip:214@1x.x.x.x:8001></a>;tag=1072578853.</span></p>
<p><span>To: <a><sip:212@x.x.x.x:8001></a>.</span></p>
<p><span>Call-ID: <a href="mailto:0_1310998066@10.0.1.111" target="_blank">0_1310998066@10.0.1.111</a>.</span></p>
<p><span>CSeq: 2 INVITE.</span></p>
<p><span>Contact:
<a><sip:214@83.x.x.x:11891;transport=TLS></a>.</span></p>
<p><span>Content-Type: application/sdp.</span></p>
<p><span>Allow: INVITE, INFO, PRACK, ACK, BYE,
CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER,
PUBLISH, UPDATE, MESSAGE.</span></p>
<p><span>Max-Forwards: 69.</span></p>
<p><span>User-Agent: Yealink SIP-T21P_E2
52.80.0.3.</span></p>
<p><span>Allow-Events:
talk,hold,conference,refer,check-sync.</span></p>
<p><span>Content-Length: 553.</span></p>
<p><span>.</span></p>
<p><span>v=0.</span></p>
<p><span>o=- 20143 20143 IN IP4 x.x.x.x.</span></p>
<p><span>s=SDP data.</span></p>
<p><span>c=IN IP4 x.x.x.x</span></p>
<p><span>t=0 0.</span></p>
<p><span>m=audio 8530 RTP/AVP 0 8 18 101.</span></p>
<p><span>a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm.</span></p>
<p><span>a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3.</span></p>
<p><span>a=crypto:3 F8_128_HMAC_SHA1_80
inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw.</span></p>
<p><span>a=rtpmap:0 PCMU/8000.</span></p>
<p><span>a=rtpmap:8 PCMA/8000.</span></p>
<p><span>a=rtpmap:18 G729/8000.</span></p>
<p><span>a=fmtp:18 annexb=no.</span></p>
<p><span>a=rtpmap:101 telephone-event/8000.</span></p>
<p><span>a=fmtp:101 0-15.</span></p>
<p><span>a=ptime:20.</span></p>
<p><span>a=sendrecv.</span></p>
<p><span>a=nortpproxy:yes</span></p>
</div>
<br>
<fieldset></fieldset>
<br>
</div></div><pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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</font></span></pre><span class="HOEnZb"><font color="#888888">
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<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
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