Thanks Jeremy. I am seeing questions similar to mine when I search - but I am not finding a solution. Was wondering if you know of something I have not found yet. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200327/92e1b497/attachment.html> -------------- next part -------------- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
Shishir Pokharel
2020-Mar-27 18:17 UTC
[asterisk-users] [Asterisk-video] gstreamer integration
You could use RTMP protocol from asterisk to rtmpsrc in gstreamer. I used chan_rtmp long time back but it only supported audio back then. From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jerry Geis Sent: Friday, March 27, 2020 8:29 AM To: Development discussion of video media support in Asterisk <asterisk-video at lists.digium.com> Subject: Re: [asterisk-users] [Asterisk-video] gstreamer integration Thanks Jeremy. I am seeing questions similar to mine when I search - but I am not finding a solution. Was wondering if you know of something I have not found yet. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200327/4f70558e/attachment-0001.html> -------------- next part -------------- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
Thanks - not familiar with chan_rtmp. Will look at it. I was also wondering about directfb ? Currently I am missing a dependency to try it. Trying to find that also for baresip. I dont see an example for "directfb" in the config file for baresip. Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200327/a449a089/attachment-0001.html> -------------- next part -------------- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video
This also seems to not be viable: "Asterisk 1.6 and 11 (help us to port it to Asterisk 13/14)" I am using asterisk 13 - RTMP does not seem to be available for that. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200327/8a8af452/attachment-0001.html> -------------- next part -------------- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video