Hi
I installed the CS MCU 3.2.1 & the SIP GW 3.2.1 on my server. I also have the latest Kamailio SIP Server v4.4.3 installed on the same server.
The SIP GW works great. I can dial from my SIP Endpoint and accept the call on the webRTC page. Audio transcode (G722) and Video transcode (H264), all is perfect.
The CS MCU works great with webRTC clients, but I cannot connect a SIP endpoint to a conference. When I try to connect my SIP Endpoint to the CS MCU, it connects, but chokes on the video and audio codec negotiation (see Attached Log).
Any help would be much appreciated!
Thanks
-VR
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Hi, Mark
For CentOS, try this link http://pan.baidu.com/s/1o7BsWlW with passcode: 21pp
Let us know whether this works for you.
Best wishes,
Zhai Lei
