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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




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