Audio Issue

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Audio Issue

Faisal Rehman
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

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Re: Audio Issue

SamyGo

Hi Faisal,
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?

What is your network topology.?
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other.

Regards,
Sammy

On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[hidden email]> wrote:
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: Audio Issue

Faisal Rehman
Hi Sammy,

I have neither changed the configuration file for any media-relay nor engaged any media proxy yet, I mean to say that configuration is totally fresh with no changes except configurations with database. The network is the simplest one as my server is in UK with a public IP running OpenSIPS on it & I just want to make pc to pc calls through it. There is no issue in call connectivity but without RTP & also running the command for RTP as tethereal -i any -R rtpevent does not show any thing or any codec flow. Yeah you are correct about the fact that changing networks can not send the RTP directly to the subscribers so is there any possible available solution for it?

 
Warmest Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: OpenSIPS users mailling list <[hidden email]>; Faisal Rehman <[hidden email]>
Sent: Tuesday, September 11, 2012 9:40 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?
What is your network topology.?
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other.
Regards,
Sammy
On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[hidden email]> wrote:
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: Audio Issue

SamyGo

Hi,
Use NAT handling for clients behind NAT and may use media relaying for those clients.
Clients will pubic IP may send/received RTP directly or go through with the media relay tool if the other end needs NAT handling.

Thanks
Sammy

On Sep 11, 2012 10:54 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

I have neither changed the configuration file for any media-relay nor engaged any media proxy yet, I mean to say that configuration is totally fresh with no changes except configurations with database. The network is the simplest one as my server is in UK with a public IP running OpenSIPS on it & I just want to make pc to pc calls through it. There is no issue in call connectivity but without RTP & also running the command for RTP as tethereal -i any -R rtpevent does not show any thing or any codec flow. Yeah you are correct about the fact that changing networks can not send the RTP directly to the subscribers so is there any possible available solution for it?

 
Warmest Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: OpenSIPS users mailling list <[hidden email]>; Faisal Rehman <[hidden email]>
Sent: Tuesday, September 11, 2012 9:40 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?
What is your network topology.?
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other.
Regards,
Sammy
On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[hidden email]> wrote:
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: Audio Issue

Faisal Rehman
Hi Sammy,

Yeah I was sure that this problem can be resolved via NAT traversal or something but I've not used it before so got to study it a bit first. What about the media relaying thing, you are talking about Media Proxy aren't you?


 
Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: Faisal Rehman <[hidden email]>
Cc: OpenSIPS users mailling list <[hidden email]>
Sent: Tuesday, September 11, 2012 11:01 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi,
Use NAT handling for clients behind NAT and may use media relaying for those clients.
Clients will pubic IP may send/received RTP directly or go through with the media relay tool if the other end needs NAT handling.
Thanks
Sammy
On Sep 11, 2012 10:54 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

I have neither changed the configuration file for any media-relay nor engaged any media proxy yet, I mean to say that configuration is totally fresh with no changes except configurations with database. The network is the simplest one as my server is in UK with a public IP running OpenSIPS on it & I just want to make pc to pc calls through it. There is no issue in call connectivity but without RTP & also running the command for RTP as tethereal -i any -R rtpevent does not show any thing or any codec flow. Yeah you are correct about the fact that changing networks can not send the RTP directly to the subscribers so is there any possible available solution for it?

 
Warmest Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: OpenSIPS users mailling list <[hidden email]>; Faisal Rehman <[hidden email]>
Sent: Tuesday, September 11, 2012 9:40 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?
What is your network topology.?
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other.
Regards,
Sammy
On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[hidden email]> wrote:
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users






_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: Audio Issue

SamyGo

Hi Faisal,

Do study Ch:9 of the opensips book. That will clear almost everything.
By media relaying I mean anything from rtpproxy or mediaproxy.

--
BR
Sammy

On Sep 12, 2012 6:51 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

Yeah I was sure that this problem can be resolved via NAT traversal or something but I've not used it before so got to study it a bit first. What about the media relaying thing, you are talking about Media Proxy aren't you?


 
Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: Faisal Rehman <[hidden email]>
Cc: OpenSIPS users mailling list <[hidden email]>
Sent: Tuesday, September 11, 2012 11:01 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi,
Use NAT handling for clients behind NAT and may use media relaying for those clients.
Clients will pubic IP may send/received RTP directly or go through with the media relay tool if the other end needs NAT handling.
Thanks
Sammy
On Sep 11, 2012 10:54 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

I have neither changed the configuration file for any media-relay nor engaged any media proxy yet, I mean to say that configuration is totally fresh with no changes except configurations with database. The network is the simplest one as my server is in UK with a public IP running OpenSIPS on it & I just want to make pc to pc calls through it. There is no issue in call connectivity but without RTP & also running the command for RTP as tethereal -i any -R rtpevent does not show any thing or any codec flow. Yeah you are correct about the fact that changing networks can not send the RTP directly to the subscribers so is there any possible available solution for it?

 
Warmest Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: OpenSIPS users mailling list <[hidden email]>; Faisal Rehman <[hidden email]>
Sent: Tuesday, September 11, 2012 9:40 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?
What is your network topology.?
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other.
Regards,
Sammy
On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[hidden email]> wrote:
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users






_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Reply | Threaded
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|

Re: Audio Issue

Faisal Rehman
Hi Sammy,

Thank you for your quickest response, yeah it is the chapter 9 of the OpenSIPS book & I'll study it but if I rely on Mediaproxy only, can that problem would be resolved?

 
Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: Faisal Rehman <[hidden email]>
Cc: OpenSIPS users mailling list <[hidden email]>
Sent: Wednesday, September 12, 2012 6:58 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
Do study Ch:9 of the opensips book. That will clear almost everything.
By media relaying I mean anything from rtpproxy or mediaproxy.
--
BR
Sammy
On Sep 12, 2012 6:51 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

Yeah I was sure that this problem can be resolved via NAT traversal or something but I've not used it before so got to study it a bit first. What about the media relaying thing, you are talking about Media Proxy aren't you?


 
Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: Faisal Rehman <[hidden email]>
Cc: OpenSIPS users mailling list <[hidden email]>
Sent: Tuesday, September 11, 2012 11:01 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi,
Use NAT handling for clients behind NAT and may use media relaying for those clients.
Clients will pubic IP may send/received RTP directly or go through with the media relay tool if the other end needs NAT handling.
Thanks
Sammy
On Sep 11, 2012 10:54 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

I have neither changed the configuration file for any media-relay nor engaged any media proxy yet, I mean to say that configuration is totally fresh with no changes except configurations with database. The network is the simplest one as my server is in UK with a public IP running OpenSIPS on it & I just want to make pc to pc calls through it. There is no issue in call connectivity but without RTP & also running the command for RTP as tethereal -i any -R rtpevent does not show any thing or any codec flow. Yeah you are correct about the fact that changing networks can not send the RTP directly to the subscribers so is there any possible available solution for it?

 
Warmest Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: OpenSIPS users mailling list <[hidden email]>; Faisal Rehman <[hidden email]>
Sent: Tuesday, September 11, 2012 9:40 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?
What is your network topology.?
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other.
Regards,
Sammy
On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[hidden email]> wrote:
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users








_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Reply | Threaded
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|

Re: Audio Issue

SamyGo
Hi again,

In any case you've to intelligently detect NAT [ hint: nat_uac_test() ] Once NAT is detected apply rtpproxy or mediaproxy to traverse audio successfully in between the two legs.

--
BR
Sammy


On Wed, Sep 12, 2012 at 7:00 PM, Faisal Rehman <[hidden email]> wrote:
Hi Sammy,

Thank you for your quickest response, yeah it is the chapter 9 of the OpenSIPS book & I'll study it but if I rely on Mediaproxy only, can that problem would be resolved?

 
Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: Faisal Rehman <[hidden email]>
Cc: OpenSIPS users mailling list <[hidden email]>
Sent: Wednesday, September 12, 2012 6:58 PM

Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
Do study Ch:9 of the opensips book. That will clear almost everything.
By media relaying I mean anything from rtpproxy or mediaproxy.
--
BR
Sammy
On Sep 12, 2012 6:51 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

Yeah I was sure that this problem can be resolved via NAT traversal or something but I've not used it before so got to study it a bit first. What about the media relaying thing, you are talking about Media Proxy aren't you?


 
Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: Faisal Rehman <[hidden email]>
Cc: OpenSIPS users mailling list <[hidden email]>
Sent: Tuesday, September 11, 2012 11:01 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi,
Use NAT handling for clients behind NAT and may use media relaying for those clients.
Clients will pubic IP may send/received RTP directly or go through with the media relay tool if the other end needs NAT handling.
Thanks
Sammy
On Sep 11, 2012 10:54 PM, "Faisal Rehman" <[hidden email]> wrote:
Hi Sammy,

I have neither changed the configuration file for any media-relay nor engaged any media proxy yet, I mean to say that configuration is totally fresh with no changes except configurations with database. The network is the simplest one as my server is in UK with a public IP running OpenSIPS on it & I just want to make pc to pc calls through it. There is no issue in call connectivity but without RTP & also running the command for RTP as tethereal -i any -R rtpevent does not show any thing or any codec flow. Yeah you are correct about the fact that changing networks can not send the RTP directly to the subscribers so is there any possible available solution for it?

 
Warmest Regards,

Faisal Rehman

From: SamyGo <[hidden email]>
To: OpenSIPS users mailling list <[hidden email]>; Faisal Rehman <[hidden email]>
Sent: Tuesday, September 11, 2012 9:40 PM
Subject: Re: [OpenSIPS-Users] Audio Issue

Hi Faisal,
What are your opensips config related to any media-relay ? Have you engaged any mediaproxy in your dialplan ?
What is your network topology.?
I can only imagine media between two endpoints getting connected directly when on same network. But when you change network the two endpoints cant possibly send RTPs directly to each other.
Regards,
Sammy
On Sep 11, 2012 8:49 PM, "Faisal Rehman" <[hidden email]> wrote:
Hello Everyone!

I have installed OpenSIPS 1.7 on my CentOS box & it is up and running fine. In the initial phase I am just testing PC to PC calls but facing a little issue with audio that is fine if we test on the same network but disappears as soon we change the network, inspite of disabling firewall on both local PC & server the issue still persists. So may I get any ideas what could be the possible reasons for this?


 
Regards,

Faisal Rehman

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users









_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users