[Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio

Sat Oct 5 00:21:18 MSD 2013

But Chrome isn't on the same network, right? Also, I'm not an expert on
this, but from what I understand, STUN binding is something that occurs
between Chrome and the media server, not a STUN server.  See Example 17
here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17.  The
STUN binding occurs between the two user agents, where one is the SIP user
and the other could be your media server.

Chrome will complain about STUN binding errors or receiving unknown
packets. If audio isn't flowing, all I'm trying to say is it might not be a
FreeSWITCH issue and you should make sure Chrome isn't the culprit before
changing too many things in FreeSWITCH. In one instance, my server's
network was the problem, and setting up the same exact FreeSWITCH (and even
Asterisk) configuration resulted in two way audio.  :D)

Not saying this is your problem, just that you should definitely be
watching what's happening in the Chrome debug logs too.

Hope this helps!

James

On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana <rafaelstnoliveira at gmail.com
> wrote:

> Thanks for the replies! I couldn't test or check anything today. As soon
> as I do the tests I will inform here the new status.
>
> @James
> My application server (nginx) is on the same network my FreeSwitch and
> Asterisk are, so I'm not using a Stun server.
>
> []'s
>
>
> 2013/10/4 James Mortensen <james.mortensen at synclio.com>
>
>> Hi Rafael,
>>
>> You didn't mention whether the server was in the cloud.  If you're server
>> is on Amazon EC2, make sure you're following the guide here:
>> https://wiki.freeswitch.org/wiki/Amazon_EC2
>>
>> Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk
>> server, do you see audio flowing?  Also, in Chrome, startup chrome from the
>> command line with the options to enable debug logging:
>>
>> chrome --enable-logging --v=11
>>
>> Then look to see if there are STUN binding errors.  Also, check
>> chrome://webrtc-internals, which will also tell you if Chrome is trying to
>> send audio.
>>
>> Is the server behind NAT or is it on the public Internet with it's own
>> public IP bound to the eth0 interface?
>>
>> Hope this helps!
>>
>>
>>
>> James
>>
>>
>>
>> On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana <
>> rafaelstnoliveira at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I'm new to telephony and FreeSwitch's world, so I apologize in advance
>>> for any nonsense I speak here.
>>>
>>> I've been trying to setup an environment where It can be possible to
>>> make a call through Google Chrome Browser using JsSIP to a standard phone
>>> device on PSTN.
>>>
>>> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't
>>> chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5
>>> instance and to get access to PSTN via this instance I had to register my
>>> Asterisk instance as a gateway on my Sofia's external profile. This part of
>>> my scenario works fine. I'm able to make calls using a softphone registered
>>> on FreeSwitch to standard phones on PSTN with no problems. What I wasn't
>>> able to do until now was the JsSIP + FreeSwitch integration.
>>>
>>> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was
>>> uncomment the line below on sip_profiles/internal.xml.
>>>
>>> <param name="ws-binding" value=":5066"/>
>>>
>>> I really don't know if just this is sufficient. Am I missing something
>>> important?
>>>
>>> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit
>>> JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a
>>> call to a PSTN phone. The connection is established but I don't get any
>>> audio in both endpoints. The same happens when I try to call the 5000 ivr
>>> extension or an user on a softphone at the same network from my Chrome
>>> browser.
>>>
>>> Assuming that all the services I've mentioned here are running on the
>>> same network, do you have any idea why I can't get audio in both endpoints
>>> of my experiment?
>>>
>>> Additional information:
>>> Ubuntu 12.04 64 bits
>>> FreeSwitch version 1.5.5 default install configuration
>>> Tryit JsSIP Demo with jssip-0.3.0.js
>>>
>>> Thanks in advance,
>>> Rafael.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
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>>> FreeSWITCH-users at lists.freeswitch.org
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>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>>
>>
>
>
> --
> Rafael Santana Oliveira
> Mestre em Ciência da Computação
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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