Did you open the websocket ports on your firewall? https://wiki.freeswitch.org/wiki/Firewall

Thanks,

Moishe Grunstein

Tornado Computer Systems, Inc.

212.400.7650 888.IPPBX.US
Service Request Email: support@nysolutions.com

Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified

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From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Iwan Budi Kusnanto
Sent: Thursday, October 03, 2013 9:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio

On Friday, October 4, 2013, Rafael Santana 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

Try to add --with-openssl when doing ./configure

Thanks in advance,

Rafael.

--
Iwan Budi Kusnanto