Adrian,

Thanks for the tip. Just to make sure I understand your comment should I do what is shown below? If I do this won't calls that didn't use mediaproxy on the original INVITE be sent through the mediproxy on re-INVITEs or is this incorrect because I didn't call fix_nated_contact() in the loose_route() section?

route { #usual sanity checks

if (loose_route()) { if (method=="INVITE") { use_media_proxy(); }; r_relay(); break; };

#ususal other processing like lookup("location") }

Regards, Paul

On Wed, 16 Mar 2005 15:39:41 +0100, Adrian Georgescu ag@ag-projects.com wrote:

Just call the use of media_proxy, if the session exists it will use it otherwise not. Simple.

Arian

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Hi All.

I'm using ser-0.9.1.

Is there a way to determine if mediaproxy is in use for an existing SIP call so that re-INVITE messages can avoid losing audio when one or the other SIP UAs are NATed?

But nothing has led me to a solution. I cannot just use lookup("location") and test the nat_flag since that won't always work on re-INVITEs. A mediaproxy function for something like is_existing_session() would be awesome to lookup the Call-ID in the existing mediaproxy sessions.

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