Re: [Freeswitch-users] Sending media streams to a media gateway
Anyone else have any ideas on this? -----Original Message----- From: fax@virgintechnologies.com [mailto:fax@virgintechnologies.com] Sent: Tuesday, February 17, 2009 05:45 PM To: freeswitch-users@lists.freeswitch.org Subject: 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 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway 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> 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 list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm mailto:MSN%3Aanthony_minessale@hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888@conference.freeswitch.org http://iax:guest@conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888@conference.freeswitch.org pstn:213-799-1400
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>wrote:
Anyone else have any ideas on this?
-----Original Message----- *From:* fax@virgintechnologies.com [mailto:fax@virgintechnologies.com] *Sent:* Tuesday, February 17, 2009 05:45 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* 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 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Sending media streams to a media gateway
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> 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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (2)
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Anthony Minessale
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Justin Miller