I just used Asterisk recently and want to use Web Call Server(WCS) for call video by SIP user Asterisk. I using WCS Flashphoner for this. I use video codec H264 only, the config for WCS and softphone is the same in all test cases.
I test with some cases:
-Call video from browser to softphone(MicroSIP): I see request INVITE (WCS) and response (softphone) packet both of them have m=video 12530 RTP/AVP 99 , a=rtpmap:99 H264/90000. I capture traffic in SIP server see that WCS sent RTP (H264) to SIP server, softphone don’t send RTP (H264) and SIP server don’t forward RTP (H264) of WCS to softphone, so I can’t see the video, audio transfered. With case call video from softphone to browser is same.
-Call video from browser to browser, it works. I can see video and audio. Inspect packet requests and respond INVITE like the first case. WCS and softphone have sent RTP (H264) to SIP server and SIP server forward them.
-Call video from softphone to softphone, it works too. It is like case browser to browser
What can happen in two cases browser to softphone and softphone to browser?
Thanks for reading.