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<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS 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
