FreeSwitch + WebRTC + JsSIP + Chrome no audio
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.
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
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
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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status. @James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server. []'s 2013/10/4 James Mortensen <james.mortensen@synclio.com>
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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server. Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D) Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too. Hope this helps! James On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana <rafaelstnoliveira@gmail.com
wrote:
Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status.
@James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server.
[]'s
2013/10/4 James Mortensen <james.mortensen@synclio.com>
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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
James, Yes, Chrome is on a machine at the same network. This log file (http://pastebin.freeswitch.org/21480) is from a test scenario where I tried, using Tryit Demo, call the 5000 extension in FS. On the FS console I can see that the ivr call follow is executed perfectly, but I can't hear anything on my headset. I got this error during this test. [11:11:1004/174001:ERROR:webrtc_audio_renderer.cc(241)] Not implemented reached in virtual void content::WebRtcAudioRenderer::Start() [11:23:1004/174001:ERROR:platform_thread_linux.cc(99)] Failed to set nice value of thread to -10 I'm checking for clues right now. Any idea about it? Thanks for the attention! 2013/10/4 James Mortensen <james.mortensen@synclio.com>
But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server.
Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D)
Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too.
Hope this helps!
James
On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < rafaelstnoliveira@gmail.com> wrote:
Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status.
@James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server.
[]'s
2013/10/4 James Mortensen <james.mortensen@synclio.com>
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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
Sounds like some part of the audio renderer is not implemented on your version of chrome? On Oct 4, 2013, at 4:58 PM, Rafael Santana <rafaelstnoliveira@gmail.com> wrote:
James,
Yes, Chrome is on a machine at the same network.
This log file (http://pastebin.freeswitch.org/21480) is from a test scenario where I tried, using Tryit Demo, call the 5000 extension in FS. On the FS console I can see that the ivr call follow is executed perfectly, but I can't hear anything on my headset. I got this error during this test.
[11:11:1004/174001:ERROR:webrtc_audio_renderer.cc(241)] Not implemented reached in virtual void content::WebRtcAudioRenderer::Start() [11:23:1004/174001:ERROR:platform_thread_linux.cc(99)] Failed to set nice value of thread to -10
I'm checking for clues right now. Any idea about it?
Thanks for the attention!
2013/10/4 James Mortensen <james.mortensen@synclio.com> But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server.
Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D)
Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too.
Hope this helps!
James
On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana <rafaelstnoliveira@gmail.com> wrote: Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status.
@James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server.
[]'s
2013/10/4 James Mortensen <james.mortensen@synclio.com> 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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Try setting the headset as the default audio device in your OS before you start chrome. Does it work on that same headset on webrtc.freeswitch.org? On Sat, Oct 5, 2013 at 10:54 AM, Michael Jerris <mike@jerris.com> wrote:
Sounds like some part of the audio renderer is not implemented on your version of chrome?
On Oct 4, 2013, at 4:58 PM, Rafael Santana <rafaelstnoliveira@gmail.com> wrote:
James,
Yes, Chrome is on a machine at the same network.
This log file (http://pastebin.freeswitch.org/21480) is from a test scenario where I tried, using Tryit Demo, call the 5000 extension in FS. On the FS console I can see that the ivr call follow is executed perfectly, but I can't hear anything on my headset. I got this error during this test.
[11:11:1004/174001:ERROR:webrtc_audio_renderer.cc(241)] Not implemented reached in virtual void content::WebRtcAudioRenderer::Start() [11:23:1004/174001:ERROR:platform_thread_linux.cc(99)] Failed to set nice value of thread to -10
I'm checking for clues right now. Any idea about it?
Thanks for the attention!
2013/10/4 James Mortensen <james.mortensen@synclio.com>
But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server.
Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D)
Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too.
Hope this helps!
James
On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < rafaelstnoliveira@gmail.com> wrote:
Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status.
@James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server.
[]'s
2013/10/4 James Mortensen <james.mortensen@synclio.com>
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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
STUN allows a client behind NAT to find the IP:port its packets are leaving externally on so that it knows the location to tell the server (FreeSWITCH) to send audio back to. In short STUN is used at whichever end is using NAT (which could be none, one or both). If FreeSWITCH is on a public IP but your PC running Chrome is on NAT (extremely likely) then Chrome still needs to use STUN. On 4 October 2013 21:21, James Mortensen <james.mortensen@synclio.com>wrote:
But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server.
Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D)
Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too.
Hope this helps!
James
On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < rafaelstnoliveira@gmail.com> wrote:
Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status.
@James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server.
[]'s
2013/10/4 James Mortensen <james.mortensen@synclio.com>
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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi guys! Thanks for your help! I got it running right now. For future reference, I did the following steps to accomplish a two way audio while calling from Chrome to a PSTN phone. I've migrated my environment to a CentOS 5.9. I've installed FreeSwitch following the default recipe, but instead of ./configure -C I did ./configure --with-openssl # suggested by @Iwan To make available websocket connection on FS, I've just uncommented the following on sip_profiles/internal.xml <!-- uncomment for sip over websocket support -->
<!--<param name="ws-binding" value=":5066"/> -->
And that is it! :D To reach PSTN I used a pre-configured Asterisk instance as gateway. It is important to note that my client (Chrome), FreeSwitch and Asterisk were all at the same network while doing this tests. PS: I turned off my iptables to eliminate any possibility of port blocking while doing my tests. Thanks again! 2013/10/6 Steven Ayre <steveayre@gmail.com>
STUN allows a client behind NAT to find the IP:port its packets are leaving externally on so that it knows the location to tell the server (FreeSWITCH) to send audio back to.
In short STUN is used at whichever end is using NAT (which could be none, one or both).
If FreeSWITCH is on a public IP but your PC running Chrome is on NAT (extremely likely) then Chrome still needs to use STUN.
On 4 October 2013 21:21, James Mortensen <james.mortensen@synclio.com>wrote:
But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server.
Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D)
Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too.
Hope this helps!
James
On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < rafaelstnoliveira@gmail.com> wrote:
Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status.
@James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server.
[]'s
2013/10/4 James Mortensen <james.mortensen@synclio.com>
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@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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Rafael Santana Oliveira Mestre em Ciência da Computação
participants (7)
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Anthony Minessale
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Iwan Budi Kusnanto
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James Mortensen
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Michael Jerris
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Moishe Grunstein
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Rafael Santana
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Steven Ayre