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List Info
Thread: R: R: Streaming multiple-bitrate
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| R: R: Streaming multiple-bitrate |

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2006-08-18 15:37:35 |
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I did not receive your server config file.
I see from the captures that you are running server version
10.0.7.2162.
Server version 10.x does not support multi-rate (rel6) .3gp.
Can you reproduce the problem with single-rate (rel5) .3gp content and
create new captures?
(Also, do you have any idea why every packet sent by your server appears
to have a bad checksum?)
-jrs
At 04:04 AM 8/18/2006, Gabriele Vinci wrote:
In attach you’ll find a zip
file with both gprs and umts network traffic log file.
Let me know
Best regards
Gabriele
Da: Jason Simpson
[real.com" eudora="autourl">
mailto:jsimpson real.com]
Inviato: giovedì 17 agosto 2006 18.29
A: Gabriele Vinci; 'Jason Simpson';
helix-server-dev helixcommunity.org
Oggetto: Re: R: [Helix-server-dev] Streaming multiple-bitrate
At 12:34 AM 8/17/2006, Gabriele Vinci wrote:
Thanks for your
interest,
in attach you’ll find the rmserver.cfg file and two rdf files.
I don’t know how to capture all activity between the phone and the
server. The helix mobile streaming server is installed on al linux Red
Hat Enterprise Linux AS release 4.
Please let me know if there is any information I can send you to
understand better this issue.
Use tcpdump on the server machine to capture the network
traffic.
Try briefly playing some content, stop playback and check the logs to
determine the handset's IP address.
Once you know the IP address of your handset, run this on the server as
root:
tcpdump -s
32000 -w gprs80kbuf.cap 'host 123.45.6.78'
replacing 123.45.6.78 with the actual IP address. Then try playing again
in the normal way, via GPRS with the 80000 byte buffer. When you are
finished playing, return to the terminal where you ran tcpdump and hit
Control-C to stop it. The network traffic that occurred between the
handset and the server should be now stored in a file named
'gprs80kbuf.cap'.
Next, switch to the UMTS handset, determine it's IP address as above,
then capture the network traffic into another capture file
('umts80kbuf.cap' for example).
It might be helpful to also make two more capture files for GPRS and UMTS
with the 120k buffer.
-jrs
Thanks again
Gabriele
Da: Jason Simpson [
real.com" eudora="autourl">
mailto:jsimpson real.com]
Inviato: giovedì 17 agosto 2006 1.06
A: Gabriele Vinci; helix-server-dev helixcommunity.org
Oggetto: Re: [Helix-server-dev] Streaming multiple-bitrate
Can you provide a little more information about your setup? Particularly
helpful would be a network capture of all activity between the phone and
the server (one for GPRS and one for UMTS -- be sure to include ALL data
between handset and server in the captures) and a copy of your
rmserver.cfg file.
-jrs
At 01:43 AM 8/16/2006, Gabriele Vinci wrote:
Hi,
Ive got a problem streaming multiple-bitrate clip (30kb-100kb) on both
gprs/umts networks.
With gprs mobile phones it works fine but with UMTS phones the
audio/video streaming is ok only for the firs 20 seconds, then the audio
starts jumping until the end of the clip.
Changing the configuration parameters inside the rdf file
(VideoPreDecoderBufferSize up to 120000) the UMTS phones are be able to
stream the audio/video properly for 40seconds but the gprs ones do not
even start the clip.
Could you please help me to understand whats happening with the audio
stream? Is it possible to configure the rdf file for a specific
user-agent profile?
Thanks,
Gabriele
Here is the rdf file Im using now.
<?xml
version="1.0"?>
<rdf:RDF
xmlns:rdf="http://www.w3.org/1999/02/22-rdf-syntax-ns"
xmlns:ccpp="
http://www.w3.org/2000/07/04-ccpp"
xmlns:prf="
http://www.wapforum.org/profiles/UAPROF/ccppschema-20010330
">
<rdf escription
ID="Profile">
<ccpp:component>
<rdf escription ID="HardwarePlatform">
<rdf:type
rdf:resource="http://www.wapforum.org/profiles/UAPROF/ccppschema-20010330#HardwarePlatform"/>
<prf:Vendor>Nokia</prf:Vendor>
</rdf escription>
</ccpp:component>
<ccpp:component>
<rdf escription ID="Streaming">
<pss5:VideoPreDecoderBufferSize>80000</pss5:VideoPreDecoderBufferSize>
<pss5:VideoDecodingByteRate>0</pss5:VideoDecodingByteRate>
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