[Freeswitch-users] Testing WebRTC with JsSIP and OverSIP

Anthony Minessale anthony.minessale at gmail.com
Wed Apr 17 06:31:59 MSD 2013


We're not ready for that yet. Try back later.
On Apr 16, 2013 7:27 PM, "Henry Huang" <red.rain.seven at gmail.com> wrote:

> Hi,
>
> Like the rest of the world, I am crazy about getting webrtc to work. Since
> FS does not yet do websocket. I have used JsSIP on Chrome and OverSIP as
> the sip proxy for websocket.
>
> I was able to call from Chrome with extension 1001 to another regular
> extension 1004 on FS with 2 way audio and dtmf working fine. The media is
> transporting directly between Chrome and FS. That is awesome.
>
> But when I try to call ext. 1001(JsSIP on Chrome) back from 1004. I am
> getting the following error from the javascript console, but I believe it's
> just reporting what the Chrome is not happy about. And if I try to pick up
> the call on Chrome, it would exit that session and kept the calling
> ringing.
>
>
>    1. JsSIP | RTC SESSION | unable to set local description
>    jssip-devel.js:3313 <http://jssip.net/download/jssip-devel.js>
>
>
>    1. Failed to parse SessionDescription. m=audio 1 RTP/SAVPF Expects at
>    least 4 fields.
>
>
>
>
>  I have checked specifically about the SAVPF field I am sending from FS,
> and it has 4 field if I am not mistaken:
>
> INVITE sip:id2uaa1s at ehg4bjsdusqf.invalid;transport=ws SIP/2.0
> Record-Route: <sip:08b1bd9007 at 72.1.46.750ddc3f-2196-1231-13a0-5254004b917b
> :10080;transport=ws;lr;ovid=f96ad63d>
> Record-Route: <sip:_NzIuMS40Ni4xMjJfNTA2MA-- at oversip-public-ip
> :5060;transport=udp;lr;ovid=f96ad63d>
> Via: SIP/2.0/WS
> oversip-public-ip:10080;branch=z9hG4bK2b52650d8dfffc0bec5baf62029018cbb31b2e6c;rport
> Via: SIP/2.0/UDP
> freeswich-public-ip;branch=z9hG4bKme6eNg3jBvHpm;received=freeswich-public-ip;rport=5060
> Max-Forwards: 10
> From: "Extension 1004" <sip:1004 at freeswich-public-ip>;tag=2SNgp6mXyc0Nr
> To: <sip:id2uaa1s at ehg4bjsdusqf.invalid;transport=ws;ov-ob=08b1bd9007>
> Call-ID: 750ddc3f-2196-1231-13a0-5254004b917b
> CSeq: 42742737 INVITE
> Contact: <sip:mod_sofia at freeswich-public-ip:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130408T160028Z~d50d43f788
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 886
> X-Fs-Support: update_display,send_info
> Remote-Party-Id: "Extension 1004" <sip:1004 at freeswich-public-ip
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1366127964 1366127965 IN IP4 freeswich-public-ip
> s=FreeSWITCH
> c=IN IP4 freeswich-public-ip
> t=0 0
> a=msid-semantic: WMS XbFAafgwqma0yucP2swUgWG29xewft33
> *m=audio 29638 RTP/SAVPF 0 8 3 13*
> a=fingerprint:sha-256
> 3A:E4:E2:49:05:98:07:F6:A4:50:EA:8E:E8:04:C7:85:9E:AF:9E:42:E0:42:D2:17:96:EA:59:B6:FA:17:C4:24
> a=rtcp-mux
> a=rtcp:29638 IN IP4 freeswich-public-ip
> a=ssrc:3312676386 cname:vzuvEASUNTLceOS7
> a=ssrc:3312676386 msid:XbFAafgwqma0yucP2swUgWG29xewft33 a0
> a=ssrc:3312676386 mslabel:XbFAafgwqma0yucP2swUgWG29xewft33
> a=ssrc:3312676386 label:XbFAafgwqma0yucP2swUgWG29xewft33a0
> a=ice-ufrag:6eSd1o51L9nRthLO
> a=ice-pwd:G5abwTWaDycHcUF9
> a=candidate:7283942946 1 udp 659136 freeswich-public-ip 29638 typ host
> generation 0
> a=candidate:7283942946 2 udp 659134 freeswich-public-ip 29638 typ host
> generation 0
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:p0HhxqXHnO/K86u3Qu1StqefUKtbbGVKA+Y8KSj1
> a=ptime:20
>
>
> At this point, I don't know how to troubleshoot further. I am posting this
> here to see if there is anyone running into similar issue and have a
> solution to this?
>
> Thanks,
>
> Henry
>
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>
> 
> 
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