[OpenSIPS-Users] One Way Audio

[duane.larson at gmail.com]

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.

On Oct 21, 2009 2:42pm, Ross Beer <ross_beer at hotmail.com> wrote:










> It looks like it is sending in to the server's IP address and back to  
> it's self which is strange.





> 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.





> If I use a different network there isn'ta problem however directly using  
> asterisk on the problem network has no issues.





> Ideally I would like to resolve this issue so all networks can use  
> OpenSips.





> I am currently testing MediaProxy however it does not appear to receive  
> the RTP stream from the soft phone either.





> Thank you for you help,





> Ross





> Date: Wed, 21 Oct 2009 17:55:10 +0000
> Subject: Re: RE: [OpenSIPS-Users] One Way Audio
> From: duane.larson at gmail.com
> To: ross_beer at hotmail.com

> 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?

> On Oct 21, 2009 11:15am, Ross Beer ross_beer at hotmail.com> wrote:
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > 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
> >
> >
> >
> >
> >
> > Date: Wed, 21 Oct 2009 15:40:38 +0000
> > Subject: Re: RE: [OpenSIPS-Users] One Way Audio
> > From: duane.larson at gmail.com
> > To: ross_beer at hotmail.com; duane.larson at gmail.com
> > CC: users at lists.opensips.org
> >
> > 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?
> >
> > On Oct 21, 2009 10:35am, Ross Beer ross_beer at hotmail.com> wrote:
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > 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:
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > 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.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > 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.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > 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.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Any advice you can offer would be appreciated.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Opensips shouldn't effect the RTP if it only load balances?
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Thanks,
> > > >
> > > >
> > > >
> > > > Ross
> > > >
> > > > Did you know you can get Messenger on your mobile? Learn more.
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> > >
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> >
> >
> >
> Stay in touch with your friends through Messenger on your mobile. Learn  
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