I will be responding to this question for community corner on todays Cluecon Weekly. Mike
On May 16, 2018, at 5:21 AM, Xenia Obolenskaya <j4v28bsjtp43hnf865@gmail.com> wrote:
recv 1815 bytes from tls/[81.23.228.150]:443 at 07:59:52.249297: ------------------------------------------------------------------------ INVITE sip:gw+sip2sip@ <><my_IP:5061;transport=tls;gw=sip2sip SIP/2.0 Record-Route: <sip:81.23.228.150:443;transport=tls;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> <> Record-Route: <sip:81.23.228.150;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> <> Record-Route: <sip:81.23.228.129;lr;ftag=KKQaHvjDvKecF;did=f94.5418aa17> <> Record-Route: <sip:81.23.228.150;lr;ftag=KKQaHvjDvKecF> <> Record-Route: <sip:85.17.186.7;r2=on;lr;ftag=KKQaHvjDvKecF> <> Record-Route: <sip:85.17.186.7;transport=tcp;r2=on;lr;ftag=KKQaHvjDvKecF> <> Via: SIP/2.0/TLS 81.23.228.150:443;branch=z9hG4bK9a88.2adb9c55.0 Via: SIP/2.0/UDP 81.23.228.129:5060;branch=z9hG4bK9a88.40d983a7.0 Via: SIP/2.0/UDP 81.23.228.150:5060;branch=z9hG4bK9a88.1adb9c55.0 Via: SIP/2.0/UDP 85.17.186.7:5060;branch=z9hG4bK9a88.80ff70a4.0;i=62a32 Via: SIP/2.0/TCP 81.23.228.160:57446;rport=55026;received=81.23.228.160;branch=z9hG4bKKBNcH3r9B48Xr Max-Forwards: 66 From: "Victor" <sip:rCodzvTEQK+uupE5QSBudQ@guest.sip2sip.info> <>;tag=KKQaHvjDvKecF To: <sip:13579@sip2sip.info> <> Call-ID: f45c2a4a-d381-1236-1298-047d7b87609e CSeq: 933861599 INVITE Contact: <sip:rCodzvTEQK+uupE5QSBudQ@81.23.228.160:55026> <> User-Agent: SylkRTC (Firefox 60.0 on Windows 8.1) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, UPDATE Content-Type: application/sdp Content-Disposition: session Content-Length: 474
v=0 o=mozilla...THIS_IS_SDPARTA-60.0 3887687583368393798 5225707675845380188 IN IP4 81.23.228.160 s=- t=0 0 a=sendrecv m=audio 54672 RTP/AVP 109 9 0 8 101 c=IN IP4 81.23.228.129 a=rtpmap:109 opus/48000/2 a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1 a=rtpmap:9 G722/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WIsk0UoOzvahjfdFEDDgAVd5hXFmUnUpvBoPK1eY
In Freeswitch debug I see the correct local SDP:
[DEBUG] mod_sofia.c:850 Local SDP sofia/sip2sip/rCodzvTEQK+uupE5QSBudQ@guest.sip2sip.info <mailto:sofia/sip2sip/rCodzvTEQK+uupE5QSBudQ@guest.sip2sip.info>: v=0 o=FreeSWITCH 1526429009 1526429010 IN IP4 <my_IP> s=FreeSWITCH c=IN IP4 <my_IP> t=0 0 m=audio 28590 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7MKm6yBppNSDw1IJDgyl0a2xkkvmashJVyTACq89
but in siptrace I see that really Freeswitch sends this strange SDP (and nothing else):
send 1529 bytes to tls/[81.23.228.150]:443 at 07:59:59.082373: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 81.23.228.150:443;branch=z9hG4bK9a88.2adb9c55.0;rport=443 Via: SIP/2.0/UDP 81.23.228.129:5060;branch=z9hG4bK9a88.40d983a7.0 Via: SIP/2.0/UDP 81.23.228.150:5060;branch=z9hG4bK9a88.1adb9c55.0 Via: SIP/2.0/UDP 85.17.186.7:5060;branch=z9hG4bK9a88.80ff70a4.0;i=62a32 Via: SIP/2.0/TCP 81.23.228.160:57446;rport=55026;received=81.23.228.160;branch=z9hG4bKKBNcH3r9B48Xr Record-Route: <sip:81.23.228.150:443;transport=tls;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> <> Record-Route: <sip:81.23.228.150;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> <> Record-Route: <sip:81.23.228.129;lr;ftag=KKQaHvjDvKecF;did=f94.5418aa17> <> Record-Route: <sip:81.23.228.150;lr;ftag=KKQaHvjDvKecF> <> Record-Route: <sip:85.17.186.7;r2=on;lr;ftag=KKQaHvjDvKecF> <> Record-Route: <sip:85.17.186.7;transport=tcp;r2=on;lr;ftag=KKQaHvjDvKecF> <> From: "Victor" <sip:rCodzvTEQK+uupE5QSBudQ@guest.sip2sip.info> <>;tag=KKQaHvjDvKecF To: <sip:13579@sip2sip.info> <>;tag=p8SyjjgjH5HZB Call-ID: f45c2a4a-d381-1236-1298-047d7b87609e CSeq: 933861599 INVITE Contact: <sip:13579@ <><my_IP>:5061;transport=tls> User-Agent: Freeswitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, outbound, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 129
v=0 o=FreeSWITCH 1526429009 1526429010 IN IP4 <my_IP> s=FreeSWITCH c=IN IP4 <my_IP> t=0 0 m=audio 0 RTP/AVP 19
Bearing in mind that sip2sip includes a=crypto attribute in RTP/AVP
I added this string in dialplan:
<action application="set" data="rtp_allow_crypto_in_avp=true"/>
othewise I always received [ERR] switch_core_media.c:4316 a=crypto in RTP/AVP, refer to rfc3711 [NOTICE] switch_channel.c:3812 Hangup sofia/sip2sip/wFxwmegJTLu4n6kK8DpXjw@guest.sip2sip.info <mailto:sofia/sip2sip/wFxwmegJTLu4n6kK8DpXjw@guest.sip2sip.info> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
But now Freeswitch doesn't send the correct SDP with RTP/SAVP (I see it just in debug).
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