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[fax at virgintechnolog...]

I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin

[anthony.minessale at g...]

you could set the variable bypass_media to true before you call bridge

<action application="set" data="bypass_media=true/>
<action application="bridge" data="sofia/internal/someuser@somehost"/>

that will negotiate a point to point media connection between the caller and callee


On Tue, Feb 17, 2009 at 5:51 PM, <fax@virgintechnologies.com (fax@virgintechnologies.com)> wrote:

[fax at virgintechnolog...]

I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message-----From: Anthony Minessale [mailto:anthony.minessale@gmail.com]Sent: Tuesday, February 17, 2009 05:34 PMTo: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] Sending media streams to a media gatewayyou could set the variable bypass_media to true before you call bridge<action application="set" data="bypass_media=true/><action application="bridge" data="sofia/internal/someuser@somehost"/>that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, <fax@virgintechnologies.com (fax@virgintechnologies.com)> wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin_______________________________________________Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)mailto:MSN%3Aanthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])IRC: mailto:sip%3A888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])mailto:googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])pstn:213-799-1400

[fax at virgintechnolog...]

Anyone else have any ideas on this? -----Original Message-----From: fax@virgintechnologies.com [mailto:fax@virgintechnologies.com]Sent: Tuesday, February 17, 2009 05:45 PMTo: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] Sending media streams to a media gateway I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message-----From: Anthony Minessale [mailto:anthony.minessale@gmail.com]Sent: Tuesday, February 17, 2009 05:34 PMTo: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] Sending media streams to a media gatewayyou could set the variable bypass_media to true before you call bridge<action application="set" data="bypass_media=true/><action application="bridge" data="sofia/internal/someuser@somehost"/>that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, <fax@virgintechnologies.com (fax@virgintechnologies.com)> wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin_______________________________________________Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)mailto:MSN%3Aanthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])IRC: mailto:sip%3A888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])mailto:googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])pstn:213-799-1400

[anthony.minessale at g...]

In that case you would need a sip proxy in place to rewrite the packets for the nat issue.
There's nothing else we can really do. We have a way to do what you want but you are using it under
circumstances we can't control.



On Wed, Feb 18, 2009 at 1:46 PM, Justin Miller <fax@virgintechnologies.com (fax@virgintechnologies.com)> wrote: