Hi there, I am running asterisk 1.8 with freepbx and asterisk now, (but have also poked around with prebuilts in ubuntu as well). And have been trying to get the endpoints and also the loopback/echo tests to work at a higher resolution than CIF.
Both of the endpoints support h264 and 1080i and asterisk has libavcodec with h264 support. Everything seems to work well including loopback, but at nasty CIF resolution (352x288). I have had a look through the sip extensions options, and mucked about changing the video bandwith setting, to no avail. I found some references circa 2005 from this list refering to implementing codec options : i.e http://markmail.org/message/xnm4jpvetvrooowt and more recently here: http://www.asteriskguru.com/archives/asterisk-dev-chansip-video-capabilities-call-bandwidth-vt62314.html I have also tried restricting formats to just h264, still works in all scenarios listed above (including using h264 in loopback test) but again at CIF only. I've also noted that there should be a passthrough mode for the sip extensions (ala freeswitch style) to let them handle negotiation, but again with no luck at getting them to run at a higher resolution. Can I force this somewhere in the config at all and if so what does the stanza look like? This is a Lab demo system and outgoing calls are restricted to voice only via trunks - and I am not trying to transcode anything. Any advise and help appreciated. Kind regards -JoelW -- _____________________________________________________________________ -- 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
