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

[Rafael Santana]

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.
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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-- 
Rafael Santana Oliveira
Mestre em Ciência da Computação
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