[Freeswitch-users] ICE SDP Params

[Brian West]

What is the call flow and devices involved? You shouldn't be getting the
webrtc SDP if your bridge line and settings are correct, can you
elaborate?  Also that second part looks like a bug those should go on JIRA.

On Thu, Sep 24, 2015 at 9:38 AM, Serge Yuriev <me at nevian.org> wrote:

> Hi,
>
> I’m trying to disable any mentions about WebRTC/DTLS in SDP. My PBX can’t
> understand them and drops calls.
>
> I use pre_answer app and get such SDP towards my PBX
>
> send 1698 bytes to udp/[10.0.64.63]:6060 at 20:02:39.595098:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300267909a88
>    From: "IT, ????? ??????" <sip:12550 at 10.0.64.63
> >;tag=357758~27154efa-6325-45a2-9e47-67e5d9302ebc-249340814
>    To: <sip:17000 at 10.0.64.100>;tag=73ccUeacZSQ7c
>    Call-ID: bb282700-5f91a0af-24648-3f40000a at 10.0.64.63
>    CSeq: 101 INVITE
>    Contact: <sip:mod_sofia at 10.0.64.100:6060>
>    RSeq: 145774175
>    User-Agent:
> FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK,
> NOTIFY
>    Require: 100rel
>    Supported: precondition, 100rel, timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 925
>
>    v=0
>    o=FreeSWITCH 1442395963 1442395964 IN IP4 10.0.64.100
>    s=FreeSWITCH
>    c=IN IP4 10.0.64.100
>    t=0 0
>    a=msid-semantic: WMS wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8
>    m=audio 26996 RTP/SAVPF 18 9 8 0 3 101 13
>    a=rtpmap:18 G729/8000
>    a=rtpmap:9 G722/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:101 telephone-event/8000
>    a=rtpmap:13 CN/8000
>    a=fingerprint:sha-256
> 09:0E:1F:D3:0D:FD:F9:54:AA:FF:AD:7F:CC:0A:85:94:D6:16:38:9F:94:43:3F:78:AC:E8:FC:8B:2B:0C:83:CA
>    a=setup:actpass
>    a=rtcp-mux
>    a=rtcp:26996 IN IP4 10.0.64.100
>    a=ssrc:1443086239 cname:huRtVlIlOmEjUjt8
>    a=ssrc:1443086239 msid:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8 a0
>    a=ssrc:1443086239 mslabel:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8
>    a=ssrc:1443086239 label:wp2SlBEpzCNtEIOknB2WO45EEsyGM2B8a0
>    a=ice-ufrag:LmUqgPUL8iiKbJJj
>    a=ice-pwd:3qCPCb0hnvq8FJ6gVQw2vhEu
>    a=candidate:7296319604 1 udp 659136 10.0.64.100 26996 typ host
> generation 0
>    a=ptime:20
>    ————————————————————————————————————
>
>
>
> If I use webrtc_enable_dtls=false, FS don’t send any codecs
>
> send 932 bytes to udp/[10.0.64.63]:6060 at 18:25:17.885114:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK311ba4379c394
>    From: "IT, ????? ??????" <sip:12550 at 10.0.64.63
> >;tag=364113~27154efa-6325-45a2-9e47-67e5d9302ebc-249450749
>    To: <sip:17000 at 10.0.64.100>;tag=FcQj96g8709vS
>    Call-ID: 4b777200-5fa1db5d-24eb6-3f40000a at 10.0.64.63
>    CSeq: 101 INVITE
>    Contact: <sip:mod_sofia at 10.0.64.100:6060>
>    RSeq: 485241663
>    User-Agent:
> FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK,
> NOTIFY
>    Require: 100rel
>    Supported: precondition, 100rel, timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 158
>
>    v=0
>    o=FreeSWITCH 1442476665 1442476666 IN IP4 10.0.64.100
>    s=FreeSWITCH
>    c=IN IP4 10.0.64.100
>    t=0 0
>    a=msid-semantic: WMS TqexzJilLgNDKOovoQ0kCWDMDNGQpja8
>    ————————————————————————————————————
>
>
> How I can get rid of ICE/WebRTC/DTLS parts only?
>
>
> --
> Serge S. Yuriev
>
>
>
>
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-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
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