[Asterisk-video] (no subject)

[Klaus Darilion]

Nico Gundacker schrieb:
> Here is a part of what eyebeam is logging. I do not know how to check the
> sending of RTP in any other way, except using some special
> package-analyzing-software, which is analyzing the hole traffic at the
> network.

Use wireshark. Get the RTP ports for audio and video from the SDP of the 
INVITE and 200 OK. Then check if there are RTP packets sent to/from the 
UDP ports used for the video stream.

regards
klaus


> 
> [08-04-30]09:14:00.624 | Info | Conferencing | "Receiving video frame on
> stream 00000047" | sua::CVideoConferencingManager::OnIncomingVideoFrame
> [08-04-30]09:14:00.624 | Warning | Conferencing | "Unable to locate stream
> 71" | sua::CVideoConferencingManager::OnIncomingVideoFrame
> [08-04-30]09:14:00.624 | Debug | Video | "Drawing I420 secondary surface" |
> sua::CDirectXVideoFrameRenderer::DrawI420orIYUVSurface
> [08-04-30]09:14:00.624 | Debug | Video | "Result of blit to primary surface
> is 0" | sua::CDirectXVideoFrameRenderer::BlitImageToPrimarySurface
> [08-04-30]09:14:00.625 | Debug | Jitter | "Packet hold time is: 41 avg
> latency: 38.9856 seq: 45686" | sua::CJitterBuffer::DeliverPacket
> [08-04-30]09:14:00.625 | Info | Audio | "Adding 160 samples to latency
> reducer" | sua::CAudioManager::DeliverIncomingData
> 
> -----Ursprüngliche Nachricht-----
> Von: asterisk-video-bounces at lists.digium.com
> [mailto:asterisk-video-bounces at lists.digium.com] Im Auftrag von Klaus
> Darilion
> Gesendet: Dienstag, 29. April 2008 16:40
> An: Development discussion of video media support in Asterisk
> Betreff: Re: [Asterisk-video] (no subject)
> 
> Nico Gundacker wrote:
>> You are right, when I use express-talk with the same videos there are no
>> such error-reports as there are shown with x-lite. But still there is no
>> video shown on both sip-phones.
> 
> Does Asterisk send video RTP packets to the SIP client?
> 
> regards
> klaus
> 
> 
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