[OpenSIPS-Users] One Way Audio

[Bogdan-Andrei Iancu]

Hi Ross,

Actually you do not need any media relay (mediaproxy or rtpproxy) here. 
As time as Asterisk is on the public side, it should directly work even 
with a natted client.

What you have to check is the SDP received by the nated client in the 
200 OK - check what IP it is instructed to send traffic to. Normally it 
should be the IP of Asterisk.

Regards,
Bogdan

Ross Beer 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't a 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.
> > > >
> > > >
> > > >
> > > Use Windows Live Messenger for free on selected mobiles. Learn more.
> > >
> > >
> > >
> > Stay in touch with your friends through Messenger on your mobile. 
> Learn more.
> >
> >
> >
> ------------------------------------------------------------------------
> Stay in touch with your friends through Messenger on your mobile. 
> Learn more. <http://clk.atdmt.com/UKM/go/174426567/direct/01/>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>