I want to know does asterisk support this:
All clients that registered in local LAN, could connect directly and their RTP set point to point. And All client that registered remotely on asterisk, could connect directly and their RTP set point to point. just only when one client from local LAN want to connect with one remotely client, the RTP could pass from Asterisk, Is this possible?
I set directmedia=nonat and nat=auto_force_rport, but no answer, and RTP bypass from asterisk and when local client call remote client, the audio is one way between them.
Who can i fix it?
I need all local client coud bypass RTP from Asterisk, And all remote client could bypass RTP from Asterisk too, but only when these two group want to call together, RTP dont bypass from asterisk.
It might work if you have directmedia=nonat, but probably won’t. It will be easier for you to try that to trawl the code to find out exactly how it handles such cases.
Thanks for your replying
Do you know which files in asterisk source are in this regards?
channels/chan_sip.c, and in later versions, channels/sip/*, as well.
Please note that these files amount to around 1.25 or more MB, with relatively few comments.