[Opalvoip-user] SRTP and SDES problem
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[Opalvoip-user] SRTP and SDES problem
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From: Etienne V. <e....@vo...> - 2017-01-18 16:52:12
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Hello, I had to accept a SIP call with sRTP with SDES method and for now it failed. I search why but I failed to see what's wrong. If I test over DTLS-SRTP, it works, but in my present case, I have to use SDES. I'm using Opal 3.16.1. Here is the INVITE I received: 2017/01/18 13:01:13.043 SIP PDU received: rem=tls$10.0.3.27:54726,local=tls$10.0.3.10:5061,if=tls$10.0.3.10:5061 INVITE sip:softphonetls2@10.0.3.10 SIP/2.0 Via: SIP/2.0/TLS 10.0.3.27:5061;branch=z9hG4bK5e8abcb2 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 582 Call-ID: 05a12cce37e94f1569d229ee0f9ce7e2@10.0.3.27:5061 To: <sip:softphonetls2@10.0.3.10> Supported: replaces, timer Contact: <sip:softphonetls@10.0.3.27:5061;transport=TLS> Max-Forwards: 70 Date: Wed, 18 Jan 2017 12:01:13 GMT From: <sip:softphonetls@10.0.3.27>;tag=as0ddd7c83 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE User-Agent: Asterisk PBX v=0 o=root 576028349 576028349 IN IP4 10.0.3.27 s=Asterisk PBX 13.9.0 c=IN IP4 10.0.3.27 t=0 0 m=audio 63730 RTP/SAVPF 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=ice-ufrag:0cea9b3a378c805d2390554a5a9532d4 a=ice-pwd:1c99a688637519eb6ff5eebd4d3d6d83 a=candidate:Ha00031b 1 UDP 2130706431 10.0.3.27 63730 typ host a=candidate:Ha00031b 2 UDP 2130706430 10.0.3.27 63731 typ host a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:UpdYDH9yHwQhR7B2LkMHdpAiSnTR1RDPqL7Haf3Q And in the log I have : 2017/01/18 13:01:13.090 RTP Session 1, cannot find info for any sender 2017/01/18 13:01:13.090 SRTP Session 1, could not add SRTP stream for SSRC=3595152654 (0xd649ad0e), no sender key (yet) 2017/01/18 13:01:13.090 RTPEp Session 1, remembering local RTP at udp$10.0.3.10:49244 on connection Call[C941a29613]-EP<sip>[fe8e2184-ba7f-571f-9110-00e04c68221c] 2017/01/18 13:01:13.090 RTP Session 1, opened: local=udp$10.0.3.10:49244-49245 remote= added default sender SSRC=3595152654 (0xd649ad0e) 2017/01/18 13:01:13.090 Media RTP Session 1, signalling set remote control address to 0.0.0.0:1, media rem= local=udp$10.0.3.10:49244, control rem=udp$0.0.0.0:1 local=udp$10.0.3.10:49245 2017/01/18 13:01:13.090 ICE Checking candidate: Host 10.0.3.27:63730 UDP component=1 priority=2130706431 foundation="Ha00031b" 2017/01/18 13:01:13.090 ICE Checking candidate: Host 10.0.3.27:63731 UDP component=2 priority=2130706430 foundation="Ha00031b" 2017/01/18 13:01:13.090 ICE RTP Session 1, ICE initial answer 2017/01/18 13:01:13.090 STUN Set credentials: username="XraMyxGdH8tl0yC2:0cea9b3a378c805d2390554a5a9532d4" password=<present> realm="" 2017/01/18 13:01:13.090 STUN Set credentials: username="0cea9b3a378c805d2390554a5a9532d4:XraMyxGdH8tl0yC2" password=<present> realm="" 2017/01/18 13:01:13.090 ICE RTP Session 1, ICE configured from remote candidates: remote-media=1 remote-control=1 2017/01/18 13:01:13.090 Media RTP Session 1, ignoring remote address as is behind NAT 2017/01/18 13:01:13.090 RTP Session 1, set remote control port: singlePortTx=false other=udp$10.0.3.27:63730 2017/01/18 13:01:13.090 Media RTP Session 1, ignoring remote address as is behind NAT 2017/01/18 13:01:13.090 Media RTP Session 1, ignoring remote address as is behind NAT 2017/01/18 13:01:13.090 RTP Session 1, set remote media port: singlePortTx=false other=udp$10.0.3.27:63731 2017/01/18 13:01:13.090 Media RTP Session 1, ignoring remote address as is behind NAT 2017/01/18 13:01:13.090 SDP-EP OnSendAnswerSDPSession - Incompatible crypto suite(s) for audio session 1 What I see is that the "ApplyCryptoKey" is called on the MediaSession instance and this method always return FALSE. Is the problem comes from the INVITE I receive, or do I have something to do to let it works ? Thanks in advance, Etienne Venot |
