Asterisk 16.9.0 with Web Call Server

Asterisk Asterisk SIP
1

Hi.

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.

2

I fixed it by changing to another video codecs from H264 to VP8. I changed the configuration at WCS using only video codecs vp8 for video calls and add it at sip.conf in SIP server. If you want to call in the opposite direction, you also need to set default codecs VP8 on the softphone (MicroSIP).
But I don’t know the reason with codecs h264 it doesn’t work in 2 of 4 test cases above.

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