jyotirmoy banerjee wrote: > Hello, Kia ora, > While trying to make a call with PJSIP we are facing an issue. > The call starts fine, but just after a couple of seconds the audio goes > blank. This is happening because there is a "in dialog" re-invite while > the call is ongoing, and this cuts the audio. > Can you please help me in pointing out what might be the cause of this > in-dialog re-invite and how to prevent this. > > PS: Before this problem I was connecting a single machine Asterisk box, > and it worked without any issue. > Now we have a 6 node mechanism behind a Kamailio. While trying to setup > a call by this mechanism we are facing this issue. It probably isn't a problem with PJSIP. In a default state Asterisk will attempt to use re-invites to have RTP streams directly go between two endpoints. Depending on network conditions this may or may not work. Do you think your topology should allow this to work? If so one way to see if it is the problem is to add directmedia=no to the respective configuration sections (or general section). This will disable the re-invites and proxy media through Asterisk. If this works then both ends can't send RTP to eachother for some reason. If you want to investigate further as you believe they should be able to then you should look at the SDP in the re-invite being sent to both sides, specifically the IP address and port and see if they are correct and go from there. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
