Hello!
I have been experimenting with drop-in replacement of old "rtpproxy-ng" module with new "rtpengine". Wondering what is wrong in my configuration: there are no security attributes in rtpengine answer on RTP/SAVPF offer. Neither "fingerprint" (DTLS) nor "crypto" (SDES).
I used Firefox 29 during this test.
1. Here's original offer :
INVITE sip:user5@..... SIP/2.0
Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 Max-Forwards: 69 To: sip:user5@...... From: "user4" sip:user4@......;tag=dvf1co8urv Call-ID: phmq9o62cv21timhfnpf CSeq: 4233 INVITE Proxy-Authorization: Digest algorithm=MD5, username="user4", realm="......", nonce="U1o8H1NaOvP3wxegsYCKOJX7S7DV/r1N", uri="sip:user5@......", response="44e7b16c55d3237f63e04b3c0b194f45" Contact: sip:user4@ ......;gr=urn:uuid:c193bcd4-aa2e-47ef-a106-22e60f5fde9e;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 607
v=0 o=Mozilla-SIPUA-29.0 15825 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:2102f082 a=ice-pwd:8733e5248a7fb087b40ea45b3ca6f634 *a=fingerprint:sha-256 32:AA:85:DB:D8:3C:E6:C3:46:07:11:9E:9F:54:B9:42:7F:5C:37:5F:9D:D1:AD:19:22:A3:AC:9C:6C:A5:A6:CD* m=audio 62290 *RTP/SAVPF* 109 0 8 101 c=IN IP4 ..... a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv .....
Here's the snippet for translation SRTP-RTP:
rtpengine_offer("force trust-address symmetric replace-origin replace-session-connection ICE=force_relay *RTP/AVP*");
2. Here's final answer (from rtpengine):
SIP/2.0 200 OK
Via: SIP/2.0/WS 9mk86fpn2d35.invalid;branch=z9hG4bK1997444 Record-Route: sip:......;lr;nat=yes Record-Route: sip:.....:15060;transport=udp;lr;ovid=3207d8cd Record-Route: sip:95f6551e81@.......:10080;transport=ws;lr;ovid=3207d8cd Contact: sip:user5@......1:49362 To: sip:user5@.....;tag=4d436110 From: "user4"sip:user4@....;tag=dvf1co8urv Call-ID: phmq9o62cv21timhfnpf CSeq: 4233 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.4 stamp 70864 Content-Length: 432
v=0 o=- 1398422264455879 3 IN IP4 ..... s=X-Lite 4 release 4.5.4 stamp 70864 c=IN IP4 ...... t=0 0 m=audio 30002 *RTP/SAVPF* 109 0 8 101 a=rtpmap:109 opus/48000/2 a=fmtp:109 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:30003 ....
Error from JsSIP:
{name: "INTERNAL_ERROR", message: "Could not negotiate answer SDP; cause =
*NO_DTLS_FINGERPRINT*", __exposedProps__: Object}
Here's the snippet for translation RTP-SRTP:
rtpengine_answer("force trust-address symmetric replace-origin replace-session-connection rtcp-mux-demux ICE=force *RTP/SAVPF* ");
There was another test with Chrome 34 with the same result.
Offer:
INVITE sip:user5@.... SIP/2.0 Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236 Max-Forwards: 69 To: sip:user5@.... From: "user4" sip:user4@....;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: 957 INVITE Proxy-Authorization: Digest algorithm=MD5, username="user4", realm=".....", nonce="U1pKZ1NaSTtLo2vjK9TGyBu6Axb+EtyN", uri="sip:user5@...", response="98f6275d3636664b611ff1411af982af" Contact: sip:user4@....;gr=urn:uuid:8cd4e797-7314-485f-b191-4a15a6581c42;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 1586 v=0 o=- 7510391807340598328 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS ErnvtgCDe9aR6LWxNRgT83r0mxtyAV87LUxT m=audio 51672 *RTP/SAVPF *111 103 104 0 8 106 105 13 126 c=IN IP4 10.61.2.151 a=rtcp:51672 IN IP4 .... ...... a=ice-ufrag:EMn7uHfSS7ulRGU2 a=ice-pwd:FskTdhj7qT6ELP7uTIb+gquQ a=ice-options:google-ice *a=fingerprint:sha-256 46:6E:E0:18:4A:C5:06:A8:26:85:ED:FE:16:C1:86:5E:8D:BC:4D:D9:F2:1A:75:81:A1:A7:CE:5A:79:4D:B7:22*a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux *a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:R6qaiU7Cm471zNF6f3Q87TyXbHjEt/VhLgUgY2ZZ a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lch+mfN/hi9QmseWu+ss1M2vA8mwRh8GQYChaJvc* a=rtpmap:111 opus/48000/2 .... Answer:
SIP/2.0 200 OK Via: SIP/2.0/WS ja9i6d3am6k8.invalid;branch=z9hG4bK6193236
Record-Route: sip:...;lr;nat=yes Record-Route: sip:....:15060;transport=udp;lr;ovid=3207d8cd Record-Route: sip:712a450958@...:10080;transport=ws;lr;ovid=3207d8cd Contact: < sip:user5@10.61.2.151:49362> To: sip:user5@....;tag=b8c8ee57 From: "user4"sip:user4@...;tag=jupqetdp1v Call-ID: 7jbhpjb4r4qt8m4s2pdb CSeq: 957 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.4 stamp 70864 Content-Length: 432 v=0 o=- 1398425917116411 3 IN IP4 ... s=X-Lite 4 release 4.5.4 stamp 70864 c=IN IP4 .... t=0 0 m=audio 30006 *RTP/SAVPF* 111 0 8 126 a=rtpmap:111 opus/48000/2 a=fmtp:111 useinbandfec=1 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-15 a=sendrecv a=rtcp:30007 ....
Error is JsSIP:
Failed to set remote answer sdp: Called with a SDP without crypto enabled
Any help on this issue would be greatly appreciated!
best regards, Alexey