[Freeswitch-users] Answering to sip2sip.info INVITE Freeswitch sends strange SDP: 'm=audio 0 RTP/AVP 19'
Michael Jerris
mike at jerris.com
Wed May 16 12:56:43 UTC 2018
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 at 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 at guest.sip2sip.info> <>;tag=KKQaHvjDvKecF
> To: <sip:13579 at sip2sip.info> <>
> Call-ID: f45c2a4a-d381-1236-1298-047d7b87609e
> CSeq: 933861599 INVITE
> Contact: <sip:rCodzvTEQK+uupE5QSBudQ at 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 at guest.sip2sip.info <mailto:sofia/sip2sip/rCodzvTEQK+uupE5QSBudQ at 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 at guest.sip2sip.info> <>;tag=KKQaHvjDvKecF
> To: <sip:13579 at 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 at guest.sip2sip.info <mailto:sofia/sip2sip/wFxwmegJTLu4n6kK8DpXjw at 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)
> _________________________________________________________________________
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