Hi,

how does sdp look like from client to kamailio and from kamailio to asterisk and backwards? I did the same and it works. Homer is very useful tool to analyze sip traffic, it helped me a lot to diagnose problems like this.

Peter

Feladó: Sajid Khan via sr-users sr-users@lists.kamailio.org Dátum: péntek, 2024. november 8. 12:42 Címzett: sr-users@lists.kamailio.org sr-users@lists.kamailio.org Másolatot kap: Sajid Khan khanasajidka85@gmail.com Tárgy: [SR-Users] Not able to use SRTP - from client to proxy Im using the module RTPengine. im having issues when ever i try to do a call using SRTP im using microsip i have checked the SDP everything ok. But when ever i place the call it gets rejected. if i disable SRTP then the calls work if (!rtpengine_offer("replace-origin replace-session-connection ICE=force RTP/SAVP RTP/AVP")) {

I only need to encrypt client -> proxy from proxy -> asterisk should be plain rtp i hope someone can help me out thank you.

The log in rtpengine shows this [1731055305.417896] ERR: [2a721ebf2c73414ab8e739d7bde912f9//1 port 14124]: [srtp] SRTP output wanted, but no crypto suite was negotiated

v=0 o=- 3940068289 3940068289 IN IP4 10.211.55.4 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/SAVP 8 0 101 c=IN IP4 10.211.55.4 b=TIAS:64000 a=rtcp:4001 IN IP4 10.211.55.4 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:456947851 cname:3b02129c45c402d7 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:4k0KSFHjmiySr1Wbwx3Ld7fL27Jit3qbF2PG0VXg a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:3OS1plXz0Vn8S0psAbCPdBXRjB31oA0bvZPPF0wF

Asterisk -> Kamailio

2024/11/08 14:29:30.425896 XX -> XX SIP/2.0 401 Unauthorized Call-ID: d43fe0X From: "XXX3" sip:10817@XX;tag=d70a113556394cde89a94ae5de8877c2 To: sip:XXXX@XXX;tag=z9hG4bKebf4.b13b9b496793da6d70decd9bc6df170d.0 CSeq: 17644 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1731076221/5c44d062c5e2eee1d666es2d2882eb06e",opaque="054b171732bc9a79",algorithm=MD5,qop="auth" Content-Length: 0 Via: SIP/2.0/UDP 10.211.55.4:64058;rport=30177;received=XXXX;branch=z9hG4bKPjf722X642729d6b814c0f8fba03

After this i get rejected from asterisk request timeout

onreply_route

# If the response contains SDP, engage RTP handling if (has_body("application/sdp")) { rtpengine_answer("replace-origin replace-session-connection RTP/AVP ICE=force"); }

invite if (has_body("application/sdp")) { # Engage RTPProxy for media handling if (!rtpengine_offer("replace-origin replace-session-connection ICE=force RTP/SAVP RTP/AVP")) { xlog("L_ERR", "RTPProxy failed to manage media for INVITE from client IP $si\n"); send_reply("500", "Internal Server Error"); exit; } else { xlog("L_INFO", "RTPProxy managed media for INVITE. SDP rewritten.\n"); } }