Hi,
20% might be true for general network usage, but not for Voice traffic.
Also it depends on the codecs used, 10% packetloss at VoIP means quality
will be degraded (codec could only compensate up to 5%).
http://www.voiptroubleshooter.com/problems/packetloss.html
g711 at 10% packetloss:
http://www.voiptroubleshooter.com/sound_files/male1_3_rep_20ms_10.wav
pre-sip / h323, but also example files at different packetloss rates,
tune in to the 20% g711 wav file... :)
http://www.networkcomputing.com/1001/1001ws2.html
Best Regards
Max M.
On 02/27/2012 03:12 PM, Faisal Rehman wrote:
Attached is the link for image:
http://i42.tinypic.com/154genn.jpg
Regards,
Faisal Rehman
------------------------------------------------------------------------
*From:* Faisal Rehman <[email protected]>
*To:* Open SIPS users mailling list <[email protected]>
*Sent:* Monday, February 27, 2012 7:02 PM
*Subject:* [OpenSIPS-Users] Packet Loss and its Solution
Hi Everyone,
I am facing huge packet loss in my server, so I am here to share with
you some of the output of packets losses that you can see in attached
image. Secondly I have few questions that I want to discuss with:
1. How can we reduce the packet loss to a minimum in an asterisk
server, I mean I just want to know more detailed info about packet
loss reduction.
2. I am calculating packet loss following that link
http://www.linuxjournal.com/article/9398 where there is written that
20% loss is acceptable, but if you see the attached image what will be
your conclusions about packet loss here?
3. At last but not least I just want to know the responsibilities of
the software & the network, I mean how much software or network is
responsible for packet loss?
4. What are the best possible ways to reduce the packet loss to a
minimum extent?
Regards,
Faisal Rehman
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