Re: DCCP voice quality experiments

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Dear Lars,
 Thanks for your reply :)..


> If you hadnt used dummy packets during
> the initial slowstart, then I guess you would have got a large packet
> loss rate! :)..

Sorry, I don't understand - there were no dummy packets.

Ok what I meant was that during the initial slowstart, the sender has
to ramp upto the encoding rate of the application. So if the
application is sending packets during that ramping period, then the
packets are going to get dropped since the transmit buffer size is
only 5. So from what I read from Vlad's thesis, I saw that he has used
idle data to achieve that appropriate rate before sending the real
data..According to his thesis,

" the initial connection establishment (for example through SIP
messaging) the two parties exchange idle data in order to achieve an
appropriate rate for beginning voice transmission."

and I presume this paper was based on Vlad's thesis work :).

But if he hadnt used idle data, then there are going to be loads of
packet losses and considering a large delay network the performance is
going to be really severe..


The configured packet loss rate that is a parameter of the experiment
causes only random drops at the router. Additional packets may be
dropped at the sender - independent of that parameter - if the apps
sends faster than the current window/rate. I believe Vlad used a send
buffer of 5 packets. If that's not in the paper, we definitely need
to add it. I also believe that send buffer overflow was extremely
rare, but Vlad would know better.

So I guess the subfactor I(e) hasnt considered the sender buffer
drops? I guess this is a major factor that needs to be considered.



Thanks again for the feedback!

You are welcome :)

--
Regards,
Arjuna

Postdoctoral Researcher
Engineering Research Lab,
Department of Engineering,
University of Aberdeen


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