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Re: [Linphone-developers] Linphone 2.99.1 Build Issue
From: |
Conrad Beckert |
Subject: |
Re: [Linphone-developers] Linphone 2.99.1 Build Issue |
Date: |
Thu, 28 Aug 2008 21:23:40 +0200 |
Hi Simon,
hey cool, it worked!!!! That's great.
I connected immediately to my Asterisk server to do some tests.
First of all, I'm impressed with the larger video output. It looks nice to get
away with the stamp size image.
I the tried to establish a connection between the Windows Version 2.99.2 and my
newly installed Linux linphone. I run into an old phenomenon which bugged me
for some time: the no video (green) at the called side (sound ok).
I Videocall
=======
I succeeded as I deactivated the h263 codec on the linux side (not the
h263-1998one). This issue is reproducable. (and a known pain from earlier
versions) The Windows Version doen't have a plain h263 codec so I couldn't test.
On the Asterisk log, I get recurring messages like this:
[Aug 28 02:53:56] NOTICE[25213]: rtp.c:1283 ast_rtp_read: Unknown RTP codec 72
received from '85.178.118.193'
[Aug 28 02:54:00] NOTICE[25213]: rtp.c:1283 ast_rtp_read: Unknown RTP codec 72
received from '85.178.118.193'
[Aug 28 02:54:05] NOTICE[25213]: rtp.c:1283 ast_rtp_read: Unknown RTP codec 72
received from '85.178.118.193'
[Aug 28 02:54:09] NOTICE[25213]: rtp.c:1283 ast_rtp_read: Unknown RTP codec 72
received from '85.178.118.193'
[Aug 28 02:54:14] NOTICE[25213]: rtp.c:1283 ast_rtp_read: Unknown RTP codec 72
received from '85.178.118.193'
sometimes I had this:
[Aug 27 22:50:30] WARNING[24417]: rtp.c:891 ast_rtcp_read: RTCP Read too short
[Aug 27 22:50:30] WARNING[24417]: rtp.c:891 ast_rtcp_read: RTCP Read too short
[Aug 27 22:50:35] WARNING[24417]: rtp.c:891 ast_rtcp_read: RTCP Read too short
[Aug 27 22:50:50] WARNING[24417]: rtp.c:891 ast_rtcp_read: RTCP Read too short
[Aug 27 22:51:05] WARNING[24417]: rtp.c:891 ast_rtcp_read: RTCP Read too short
The linux version constantly prints into its terminal while a videocall is
active:
[h263 @ 0xb76159a8]warning, clipping 1 dct coefficient to -127 .. 127
[h263 @ 0xb76159a8]warning, clipping 1 dct coefficient to -127 .. 127
...
and then
[h263 @ 0xb76159a8]vbv buffer overflow
The image quality from the Linux side is somewhat poor. (blocks, artefacts
etc.) The Windows side sends better images (aarrggh?? Why??) Bandwith is set
to 512/192 (which I don't have since I use up my bandwith by sending the rtp
data back and forth to my asterisk server at a different location)
The difference in quality is striking.
II Audio
=====
Anyway. I managed to get a connection using the gsm audio codecs. The speex one
cause crackling noise. (that can be an issue with my notebook though)
There is one phenomenon I had in production - and managed to get in my tests as
well (not reproducable unfortunately): Sound is OK, one can hear background
noise from the other side crystal clear - but once the speaker starts to talk
the volume goes zero (and hence one cannot understand what the other person
says)
III x264
=====
I then tried to compile the x264 codec. That succeeds but I cannot see it in
the list of codecs in linphone. I then deleted everything that looks like 264 -
with the result to kill ffmpeg. (linphone wouldn't start anymore) Anyway. What
am I supposed to to to get x264 to work? (parameters etc)
Gosh - its late.
Hope I could help you improving. Please give me a hint what I should test next.
Greetings
Conrad