discussion Digest, Vol 34, Issue 3

Send discussion mailing list submissions to
	discussionsipforum.org

To subscribe or unsubscribe via the World Wide Web, visit
	http:
//sipforum.org/mailman/listinfo/discussion
or, via email, send a message with subject or body 'help'
to
	discussion-requestsipforum.org

You can reach the person managing the list at
	discussion-ownersipforum.org

When replying, please edit your Subject line so it is more
specific
than "Re: Contents of discussion digest..."


Today's Topics:

   1. hi jitter value.. (priyaranjan das)
   2. Re: hi jitter value.. (Raj)


------------------------------------------------------------
----------

Message: 1
Date: Fri, 2 May 2008 16:18:10 +0530
From: "priyaranjan das" <priyaranjan81gmail.com>
Subject: [SIPForum-discussion] hi jitter value..
To: discussionsipforum.org
Message-ID:
	<25149d710805020348g67a37d4bsefa80d025f87de82mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

In a SIP network, I have faced a very piculiar problem.While
I am in a call
and I want to see the network parameter in my
phone,sometimes the jitter
value in my phone shows 70msc or even 80 msc.At that same
time the oneway
network delay varies from 11000msc to -10546msc.But still
they dont have any
effect on the voice quqlity.

Can u suggest me in which case the jitter value and the
network delay time
rises to that extent,without affecting the voice quqlity.

-- 
Thanks & Regards,
Priyaranjan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://sipforum.org/p
ipermail/discussion/attachments/20080502/811220dd/attachment
-0001.html 

------------------------------

Message: 2
Date: Fri, 2 May 2008 17:54:32 +0530
From: Raj <rajasekhar.lgmail.com>
Subject: Re: [SIPForum-discussion] hi jitter value..
To: "priyaranjan das" <priyaranjan81gmail.com>
Cc: discussionsipforum.org
Message-ID:
	<24bc18350805020524w1c896892yc39940e60b4518b0mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,
i am not sure about which algorithm you are following for
jitter buffer. The
network delay is variable one, most of the jitter buffer
algorithms will
calculate the jitter value using the estimated network delay
and the fixed
delay(delay for processing the vocie packet after receiving
from network).
The jitter buffer will buffer the voice packets and it will
play at the end
point to make the listener not to feel uncomfortable because
of these
delays.




On 5/2/08, priyaranjan das <priyaranjan81gmail.com> wrote:
>
> Hi,
>
> In a SIP network, I have faced a very piculiar
problem.While I am in a
> call and I want to see the network parameter in my
phone,sometimes the
> jitter value in my phone shows 70msc or even 80 msc.At
that same time the
> oneway network delay varies from 11000msc to
-10546msc.But still they dont
> have any effect on the voice quqlity.
>
> Can u suggest me in which case the jitter value and the
network delay time
> rises to that extent,without affecting the voice
quqlity.
>
> --
> Thanks & Regards,
> Priyaranjan
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please
visit
> http:
//sipforum.org/mailman/listinfo/discussion
> Post to the list at discussionsipforum.org
>
>


-- 
-Raaz
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://sipforum.org/p
ipermail/discussion/attachments/20080502/777713aa/attachment
-0001.html 

------------------------------

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http:
//sipforum.org/mailman/listinfo/discussion
Post to the list at discussionsipforum.org


End of discussion Digest, Vol 34, Issue 3
*****************************************