We have tried the Intel CS for WebRTC conference with our legacy SIP server and works seamlessly. Please let us know more details requested below:
Thanks in Advance for providing the required details.
Nice to hear you are building produce with Intel CS for WebRTC. For the questions you concern, see my answers as following,
1. Yes. Free to use and redistribute.
2. Currently not an open source project.
3. Currently no commercial version, we are also working on the supporting model evolution, will update on forum if any change. But welcome to share your story with Intel CS for WebRTC team through email@example.com.
4. We support Opus, and support customer to integrate AAC codec into Intel CS for WebRTC. Details please see server installation guide.
5. Currently 1080p is well tested.
Thank you for the response.
We were evaluating the Intel CS for WebRTC for our MCU requirement and we observed the following issue with SIP agent
There is jump in SIP packet captured in PC with Intel CS(SIP agent) from 16:16:58 to 16:17:14 – almost 16 seconds, no SIP packets captured in PC even though, remote SIP endpoint kept sending the SIP messages like INVITE (though other rtp packets/ipv6 packets are received in PC with SIP agent, during this time). During this time, remote SIP endpoint received ICMP Destination unreachable response.
Hence I suspect in PC with Intel MCU, sip Agent wasn’t responsive or crashed during this interval. We need support from Intel MCU team to further investigate this issue.
Logs around this time:
2017-01-24 16:13:18.531 - INFO: ErizoAgent - message from worker ee94140e-b3a7-e146-2180-a2fa6ca36ce0 : READY
2017-01-24 16:17:04.566 - INFO: ErizoAgent - ce2d2062-eed5-e66e-58f8-49f5ddc0dcd8 exit with null
2017-01-24 16:13:18.373 - INFO: SipPortal - Sip node init successfully.
2017-01-24 16:17:15.656 - INFO: SipErizoHelper - ErizoJS[ ce2d2062-eed5-e66e-58f8-49f5ddc0dcd8 ] check alive timeout!
Thanks in Advance,
The real sip terminal for MCU part is sip node, not sip agent. Sip agent is responsible to spawn sip node to serve on room with sip connectivity configuration. You need sip server and configure one sip account for one room. Details please refer conference server user guide.