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

[James Mortensen]

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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