[Freeswitch-users] WebRTC Call

Wed Dec 12 11:31:34 MSK 2012

If you are using newer (doubango based) version of WebRTC2SIP gateway then
you don't need to do anything, it will take care of both signalling and
media. However you might need media breaker module for transcoding. Though
FS supports both ISAC and VP8 codecs (for audio and video respectively) but
i am not sure if they are compatible / working with Chrome, since chrome
has lot of development going around and they make so many changes on daily
basis, resulting in every new version pretty much incompatible with older
ones and so on.

Please make sure you follow this wiki,

http://code.google.com/p/webrtc2sip/wiki/Building_Source_v2_0

Thank you.

On Wed, Dec 12, 2012 at 4:04 AM, Joel Rosenfield
<joelrosenfield at yahoo.com>wrote:

> In the default FreeSWITCH configuration, I used the webrtc2sip proxy to
> connect a Chrome 23 client to user 1001.  The SIP signaling is fine for
> REGISTER and INVITE, however FreeSWITCH returns 488 Not Acceptable Here.
> I saw a post on the discuss-doubango<https://groups.google.com/forum/#!forum/doubango> Google
> Groups list from July 20 that said:
>
> "The problem is not with the crypto its with the a=crypto being inside
> the AVP vs SAVP (denoting secure)
> see http://jira.freeswitch. org/browse/FS-636<http://jira.freeswitch.org/browse/FS-636>
>
> I am willing to lift this restriction or at least make it configurable
> since the alternative is you must send a double sized sdp with AVP for non
> secure stuff and SAVP for the secure."
>
> Is there something that I need to configure on FreeSWITCH to remove this
> restriction?  Or, what should be different in the SDP offer from webrtc2sip
> so that FreeSWITCH will accept it?
> Below is the output from the FreeSWITCH version command, the SDP offer
> from webrtc2sip, and FreeSWITCH error message.
>
> Thanks,
> - Joel
>
> version
>
> FreeSWITCH Version 1.3.8b+git~20121205T191750Z~ 924c524197 (git 924c524
> 2012-12-05 19:17:50Z)
> . . . .
>    ----------------------------- ------------------------------
> -------------
> recv 1931 bytes from udp/[10.159.25.56]:10060 at 22:42:51.414083:
>    ----------------------------- ------------------------------
> -------------
>    INVITE sip:9195@ . . .
>     . . .
>    User-Agent: webrtc2sip Media Server 2.0
>    P-Preferred-Identity: <sip:webrtc2sip>
>
>    v=0
>    o=doubango 1983 678901 IN IP4 10.159.25.56
>    s=-
>    c=IN IP4 10.159.25.56
>    t=0 0
>    m=audio 60326 RTP/AVP 0 8 101
>    c=IN IP4 10.159.25.56
>    a=ptime:20
>    a=silenceSupp:off - - - -
>    a=rtpmap:0 PCMU/8000/1
>    a=rtpmap:8 PCMA/8000/1
>    a=rtpmap:101 telephone-event/8000/1
>    a=fmtp:101 0-16
>    a=tcap:1 RTP/SAVP
>    a=pcfg:1 t=1
>    a=sendrecv
>    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:
> CG0ol5gUQjNxOXyMhXSIV2RltFltbx 99IkjHmXsT
>    a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:aiZ33ixxaTy5dX5iy72n/
> OAORs8lIezYKhuzti6G
>    a=rtcp-mux
>    a=ssrc:2783445458 cname:ldjWoB60jbyQlR6e
>    a=ssrc:2783445458 mslabel:6994f7d1-6ce9-4fbd- acfd-84e5131ca2e2
>    a=ssrc:2783445458 label:Doubango
>    a=ice-ufrag:kWutfVwWe7bWeB4
>    a=ice-pwd: ziAr5WnKhzu1KsZsMinJw
>    a=mid:audio
>    a=candidate:35wALsfKp 1 udp 2130706431 10.159.25.56 60326 typ host
>    a=candidate:35wALsfKp 2 udp 2130706430 10.159.25.56 60327 typ host
>    a=candidate:srflx35wA 2 udp 1694498814 107.21.197.144 60327 typ srflx
>    a=candidate:srflx35wA 1 udp 1694498815 107.21.197.144 60326 typ srflx
>    ----------------------------- ------------------------------
> -------------
> send 400 bytes to udp/[10.159.25.56]:10060 at 22:42:51.414473:
>    ----------------------------- ------------------------------
> -------------
>    SIP/2.0 100 Trying
>    . . .
>    ----------------------------- ------------------------------
> -------------
> 2012-12-11 22:42:51.734425 [INFO] mod_dialplan_xml.c:498 Processing 1001
> <1001>->9195 in context default
> -> 2012-12-11 22:42:51.894429 [ERR] sofia_glue.c:4922 a=crypto in RTP/AVP,
> refer to rfc3711
> 2012-12-11 22:42:51.894429 [NOTICE] switch_channel.c:3484 Hangup
> sofia/internal/1001 at . . .
>  [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> send 924 bytes to udp/[10.159.25.56]:10060 at 22:42:52.054746:
>    ----------------------------- ------------------------------
> -------------
>    SIP/2.0 488 Not Acceptable Here
>    Via: SIP/2.0/UDP 10.159.25.56:10060;branch=
> z9hG4bK1356314285436;rport= 10060
>
> NOTE:  I hacked the webrtc2sip to send "a=tcap:1 RTP/SAVP" rather than
> "a=tcap:1 RTP/SAVPF", but as you can see, no luck; same result either way.
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>

-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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