<div dir="ltr">Hi Rafael,<div><br></div><div>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: <a href="https://wiki.freeswitch.org/wiki/Amazon_EC2">https://wiki.freeswitch.org/wiki/Amazon_EC2</a></div>
<div><br></div><div>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:</div>
<div><br></div><div>chrome --enable-logging --v=11</div><div><br></div><div>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.</div>
<div><br></div><div>Is the server behind NAT or is it on the public Internet with it's own public IP bound to the eth0 interface? </div><div><br></div><div>Hope this helps!</div><div><br></div><div class="gmail_extra">
<br clear="all"><div><div dir="ltr"><div><br></div><div>James<br></div><div><br><div><br></div><div><br></div><div>On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana <span dir="ltr"><<a href="mailto:rafaelstnoliveira@gmail.com" target="_blank">rafaelstnoliveira@gmail.com</a>></span> wrote:<br>
</div></div></div></div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>I'm new to telephony and FreeSwitch's world, so I apologize in advance for any nonsense I speak here. </div>
<div><br></div><div>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. </div>
<div><br></div><div>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. </div>
<div><br></div><div>To setup FreeSwitch to comunicate with JsSIP, the only thing I did was uncomment the line below on sip_profiles/internal.xml.</div><div><br></div><div><param name="ws-binding" value=":5066"/></div>
<div><br></div><div>I really don't know if just this is sufficient. Am I missing something important?<br></div><div><br></div><div>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.</div>
<div><br></div><div>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?</div><div><br></div><div>Additional information:</div>
<div>Ubuntu 12.04 64 bits</div><div><div>FreeSwitch version 1.5.5 default install configuration</div><div>Tryit JsSIP Demo with jssip-0.3.0.js</div><div><br></div>Thanks in advance,</div><div>Rafael.</div></div>
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