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 reasonDid 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.
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
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