[asterisk-users] End to End delay calculation
capricorn 80
cool_capricorn80 at hotmail.com
Sun Nov 22 19:44:16 CST 2009
Hi ! Yea lot of things to look but what in case of sip phone to sip phone ? Is there anyway we can do it with some open source tool ? I have to do it for my experiment and I am really worried about it.
Regards,
> Date: Sun, 22 Nov 2009 17:12:22 -0800
> From: asterisk.org at sedwards.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] End to End delay calculation
>
> On Sun, 22 Nov 2009, capricorn 80 wrote:
>
> > I am looking to calculate the end-to-end delay between two soft
> > phone/hard phone. I have asterisk server and configured ntp server on
> > the same machine and synchronized it with ntp pool. I have seen that
> > Wireshark can be used to check the jitter. But I am not sure how can i
> > calculate the end to end. May be this is not related to the mailing list
> > topic but please help me if anyone has some information.
>
> A very long time ago, I made the mistake of letting a client listen (with
> a handset on each side of his head) to end-to-end delay.
>
> This all of a sudden became a quest for the Holy Grail to quantify and
> reduce the delay.
>
> I got a couple of RadioShack telephone recording interfaces, connected one
> to each endpoint. Then I connected the outputs to the left and right
> channels on a PC and recorded "tapping" on one of the handsets using
> Audacity. When I selected the interval between the "tap" and the "ping,"
> Audacity would show the time in ms.
>
> All very "old-school" but it worked and the client never questioned the
> "pretty pictures" on the computer screen.
>
> Wireshark may be able to tell you how long it takes a packet to travel
> across your network, but what about the time from the network interface on
> the host until sound comes out the earpiece? How long does it take a SIP
> phone to take a packet off it's network interface, wiggle it through it's
> jitter buffer, transcode it, convert it to analog and deliver it to the
> earpiece?
>
> --
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
_________________________________________________________________
Keep your friends updated—even when you’re not signed in.
http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091123/a9d63ca5/attachment.htm
More information about the asterisk-users
mailing list