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:
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 [CS_EXECUTE]
[INCOMPATIBLE_DESTINATION]
But now Freeswitch doesn't send the correct SDP with RTP/SAVP (I see it
just in debug).
FreeSWITCH Version 1.6.19+git~20171120T163416Z~b1b21d0695~64bit (git
b1b21d0 2017-11-20 16:34:16Z 64bit)