Hi, Thanks for the idea about packet compression. By 'call fails to connect', I meant the call does not connect to the callee, ie. the callee's phone does not ring after the INVITE (despite using TURN server).
This was a public WiFi network and that was all I could get at the time. I am using OpenSIPS version 2.1. Nabeel On 6 May 2016 9:16 am, "Bogdan-Andrei Iancu" <[email protected]> wrote: > Hi, > > Hard to analyze a call based on the INVITE packet only :). Still the SIP > signaling does not show any ALG interference (also not sure if the capture > was done before or after the ALG). Also, what you mean by "call fails" ?no > reply, negative reply , no audio ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 05.05.2016 22:35, Nabeel wrote: > > > Please check the following SIP trace taken within a WiFi network. The call > fails to connect despite the INVITE request and using a non-standard port. > Could this be caused by SIP ALG, or some unopened RTP port on the router? > > http://pastebin.com/raw/C4iymTbh > > > _______________________________________________ > Users mailing > [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >
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