[OpenSIPS-Users] One Way Audio

[Ross Beer]

Yep, traffic comes from the asterisk server and can be heard on the softphone, but when the echo test starts no audo can be heard.
 
Therfore the flow goes like this:
 
Asterisk ---> Opensips ----> Softphone
 
But NOT:
 
Softphone ---> Opensips ----> Asterisk
 
Which is strange, if opensips is not in the path all works correctly. Also if I call out using a SIP provider I also get two way audio, but not when talking directly to asterisk.
 
Regards,
 
Ross
 


From: ross_beer at hotmail.com
To: duane.larson at gmail.com; users at lists.opensips.org
Subject: RE: [OpenSIPS-Users] One Way Audio
Date: Wed, 21 Oct 2009 16:35:28 +0100



NHi Duane,
 
There are is a firewall on the server end however all ports are open, no NAT at the server end however there is NATing on the end of the soft phone. Though when registering with asterisk directly there is no issue.
 
Regards,
 
Ross
 


Date: Wed, 21 Oct 2009 15:23:04 +0000
Subject: Re: [OpenSIPS-Users] One Way Audio
From: duane.larson at gmail.com
To: ross_beer at hotmail.com

Are there any firewalls or NATing involved? 

On Oct 21, 2009 10:13am, Ross Beer <ross_beer at hotmail.com> wrote: 
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> I have a server located on the internet running opensips and asterisk. When registering directly to asterisk I can perform echo tests and make calls. 
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> If I register to Opensips and use the load_balance there is one way audio. I can hear sounds coming from the asterisk server but sound from the soft phone does not reach asterisk. I can confirm this when looking at a rtp debug on asterisk. 
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> I can see that traffic is passing from the soft phone when performing a wire shark trace to the server and it also shows that some RTP packet are being passed out and back into my local address. This does not happen if I register directly to asterisk. 
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> Any advice you can offer would be appreciated. 
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> Opensips shouldn't effect the RTP if it only load balances? 
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> Thanks, 
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> Ross 
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