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

[Rafael Santana]

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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