Now that you are testing with Mediaproxy are you sure that you are using the use_media_proxy() correctly? The more info you provide the more we can help. SIP traces are good.
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<br />On Oct 21, 2009 2:42pm, Ross Beer <ross_beer@hotmail.com> wrote:
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<br />> It looks like it is sending in to the server's IP address and back to it's self which is strange.
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<br />> I think this has something to do with the SDP and possibly my router. I am doing an echo test so audio should come back, however Asterisk should stay in the media path as it does when directly using asterisk.
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<br />> If I use a different network there isn't a problem however directly using asterisk on the problem network has no issues.
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<br />> Ideally I would like to resolve this issue so all networks can use OpenSips.
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<br />> I am currently testing MediaProxy however it does not appear to receive the RTP stream from the soft phone either.
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<br />> Thank you for you help,
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<br />> Ross
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<br />> Date: Wed, 21 Oct 2009 17:55:10 +0000
<br />> Subject: Re: RE: [OpenSIPS-Users] One Way Audio
<br />> From: duane.larson@gmail.com
<br />> To: ross_beer@hotmail.com
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<br />> In the wireshark trace what IP is the softphone sending the RTP packets to? Whats the destination? Is it actually sending the RTP to the Asterisk box?
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<br />> On Oct 21, 2009 11:15am, Ross Beer ross_beer@hotmail.com> wrote:
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<br />> > 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.
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<br />> > Therfore the flow goes like this:
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<br />> > Asterisk ---> Opensips ----> Softphone
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<br />> > But NOT:
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<br />> > Softphone ---> Opensips ----> Asterisk
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<br />> > 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.
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<br />> > Regards,
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<br />> > Ross
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<br />> > Date: Wed, 21 Oct 2009 15:40:38 +0000
<br />> > Subject: Re: RE: [OpenSIPS-Users] One Way Audio
<br />> > From: duane.larson@gmail.com
<br />> > To: ross_beer@hotmail.com; duane.larson@gmail.com
<br />> > CC: users@lists.opensips.org
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<br />> > So in the wireshark trace you see RTP traffic coming from the Asterisk servers IP address, but what about the traffic coming from the softphone? What IP address is that going towards?
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<br />> > On Oct 21, 2009 10:35am, Ross Beer ross_beer@hotmail.com> wrote:
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<br />> > > NHi Duane,
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<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.
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<br />> > > Regards,
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<br />> > > Ross
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<br />> > > 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
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<br />> > > Are there any firewalls or NATing involved?
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<br />> > > On Oct 21, 2009 10:13am, Ross Beer ross_beer@hotmail.com> wrote:
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<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.
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<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.
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<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.
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<br />> > > > Any advice you can offer would be appreciated.
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<br />> > > > Opensips shouldn't effect the RTP if it only load balances?
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<br />> > > > Thanks,
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<br />> > > > Ross
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