acappel01 / sipml5

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One way voice in inbound call to sipml5 client from FreeSWITCH server due to DTLS #126

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?
1. Latest FreeSWITCH, Latest SIPml5 code, OpenSIPS Server (Registrar Server) 
and OverSIP 1.4.0 (WebRTC SIP Proxy)
2. When make an inbound call to sipml5 and bridge it to a conference on 
freeswitch, there is no music heard which we have set.
3. Whereas when we make an outbound call from sipml5 to freeswitch, when it 
goes into the conference, we can hear the music.
4. When we disable the dtls thing on the freeswitch, we can hear the music in 
the inbound scenario too.
5.) Whereas when we don't disable the dtls and use JsSIP instead, we can hear 
the music

What is the expected output? What do you see instead?
Ans: We should be able to hear the conference music without disabling the dtls 
on sipml5

What version of the product are you using? On what operating system?

SIPml5:

Path: .
URL: http://sipml5.googlecode.com/svn/trunk
Repository Root: http://sipml5.googlecode.com/svn
Repository UUID: 63d8a0da-676c-2731-e3aa-b417aa27da68
Revision: 213
Node Kind: directory
Schedule: normal
Last Changed Author: bossiel@yahoo.fr
Last Changed Rev: 213
Last Changed Date: 2013-08-10 14:05:40 +0000 (Sat, 10 Aug 2013)

FreeSWITCH:

a4a0c3f (git version)

OpenSIPS:

1.9.1

OverSIP:

1.4.0

Please provide any additional information below.

Command from FreeSWITCH to call sipml5 client:

originate {media_webrtc=true,absolute_codec_string=OPUS}sofia/sip/9003@a.b.c.d 
&conference(9003)

Logs have been attached for SIPml5 console. The public IP of our server has 
been replaced by a.b.c.d in the log file attached.

All the components are on the same server i.e. SIPml5, OpenSIPS, FreeSWITCH and 
OverSIP and even JsSIP.

One thing that we have noticed is when the call is made on the JsSIP client the 
DTLS gets activated and the voice comes after that and there is no DTLS thing 
happening in the SIPml5 code.

Original issue reported on code.google.com by aami...@gmail.com on 3 Oct 2013 at 9:12

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