If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. This message may contain information that is not intended for you. De Staat aanvaardt geen aansprakelijkheid voor schade, van welke aard ook, die verband houdt met risico's verbonden aan het electronisch verzenden van berichten. Indien u niet de geadresseerde bent of dit bericht abusievelijk aan u is toegezonden, wordt u verzocht dat aan de afzender te melden en het bericht te verwijderen. I've also setup the videobridge with openfire and that seemed to work well, although I only tested a little bit.ĭit bericht kan informatie bevatten die niet voor u is bestemd. Even ZRTP worked well in both audio and video.īefore I start sending logs, etc, I was wondering if anyone has been able to use Jitsi's video with SIP, and with Freeswitch in particular.
This time video worked as it should (most of the time, I still get inconsistent results). On the second try Jitsi is able to reach the server as it should, although video still doesn't work.įrustrated with this SIP experiment and Freeswitch, I set up an XMPP server (openfire) and tried the same experiment. Also, I often have to hangup the first attempt and try again since jitsi seems to hang after "dialing" (and I see in the server that it never receives anything from jitsi). I've tried playing around with VP8 and H264, to no avail.
Basically the video box remains greyed out on both sides during the call, and there seems to be no video media in the call (although audio works). This should theoretically let the 2 jitsi endpoints talk to each other directly with little interference from the server. I'm using Freeswitch as the SIP server and have enabled the "proxy media" variable in the SIP profile. I've been trying to get video working on a basic SIP call between 2 jitsi (latest nightly) clients. Onderwerp: Video Call through SIP (Freeswitch) not working With xmpp nothing is inbetween, which can be an advantage, but also a serious limitation. *even difference between client1 calling client2 and client2 calling client1 Video-codec mismatch between client1/pbx and client2/pbx, or client1/client2. I've seen same behaviour* as you, but mostly (!) It boils down, to. Yes, i've experimenting with sip-video as well, but with asterisk. (Sorry 4 topposting, forced upon me by BB)
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