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NHi Duane,<BR>
<BR>
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.<BR>
<BR>
Regards,<BR>
<BR>
Ross<BR> <BR>
<HR id=stopSpelling>
Date: Wed, 21 Oct 2009 15:23:04 +0000<BR>Subject: Re: [OpenSIPS-Users] One Way Audio<BR>From: duane.larson@gmail.com<BR>To: ross_beer@hotmail.com<BR><BR>Are there any firewalls or NATing involved? <BR><BR>On Oct 21, 2009 10:13am, Ross Beer <ross_beer@hotmail.com> wrote: <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> 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. <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> 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. <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> 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. <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> Any advice you can offer would be appreciated. <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> Opensips shouldn't effect the RTP if it only load balances? <BR>> <BR>> <BR>> <BR>> <BR>> <BR>> Thanks, <BR>> <BR>> <BR>> <BR>> Ross <BR>> <BR>> Did you know you can get Messenger on your mobile? Learn more. <BR>> <BR>> <BR>> <br /><hr />Use Windows Live Messenger for free on selected mobiles. <a href='http://clk.atdmt.com/UKM/go/174426567/direct/01/' target='_new'>Learn more.</a></body>
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