Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: water235 on March 22, 2017, 05:28:29 PM
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I have a SIP Server -
I have a front end WebRTC(not genesys) send a Invite for a video call to SIP Server .
The call reaches the SIP RP. i could see the request gets to RM and then TO MCP.
I see both the audio and video and SIP doesn't cut the video because i have disabled the SIP-filter-media. so it permits video.
I see the voice channel getting established - i.e i could see the sound getting played, but video doens't any additional tweaking required for SIP Video ?
I haven't modified anything else on the SIP server for this video..
Any thoughts or pointers is highly appreciated.
Thanks
Water
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WebRTC is tricky and still on diapers. Check the browser versions and see if they support your WebRTC implementation. I once had to downgrade for it to work until WebRTC libraries were updated.
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