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,