[Freeswitch-users] Testing WebRTC with JsSIP and OverSIP

Henry Huang red.rain.seven at gmail.com
Wed Apr 17 04:23:41 MSD 2013


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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