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Wireshark-users: Re: [Wireshark-users] End to End VoIP delay calculation (Interarrival jitter)

From: capricorn 80 <cool_capricorn80@xxxxxxxxxxx>
Date: Fri, 27 Nov 2009 23:17:10 +0000

 HI !

  I have checked that and didn't pay mention attention to it but now I have downloaded the aaa.pcap and working it. In this file all communication is from

Sender: 192.168.1.2 to Destination: 212.242.33.36

but in my case on 31 I am getting source 61.216.159 and destination 113.

31 24.685764 61.216.159.110    113.100.26.222 RTP     0.009784 
 
Why its like that in my case ?

Regards,


> Date: Fri, 27 Nov 2009 16:11:29 +0100
> From: Lars.Ruoff@xxxxxxxxxxxxxxxxxx
> To: wireshark-users@xxxxxxxxxxxxx
> Subject: Re: [Wireshark-users] End to End VoIP delay calculation (Interarrival jitter)
>
> Have you checked http://wiki.wireshark.org/RTP_statistics => How jitter
> is calculated ?
>
> Regards,
> Lars
>
> ________________________________
>
> From: wireshark-users-bounces@xxxxxxxxxxxxx
> [mailto:wireshark-users-bounces@xxxxxxxxxxxxx] On Behalf Of capricorn 80
> Sent: vendredi 27 novembre 2009 15:44
> To: wireshark-users@xxxxxxxxxxxxx
> Subject: Re: [Wireshark-users] End to End VoIP delay calculation
> (Interarrival jitter)
>
>
>
> Hi !
>
> Thanks for your responses. @ martin.r.mathieson: I tried alot
> to understand but may be I dont have much expertise in this case :(.
> .Now I am doing like this that I have run wireshark on
> computer and computer is synchronized with ntp server. I am looking for
> interarrival calculation.
>
> This is my readings from wireshark: (The IP addresses i
> mentioned is dummy one).
>
> 113.100.26.222 is computer
> 61.216.159.110 is asterisk server
>
> No Time Source Destination
> Protocol Delta time
>
> ------------------------------------------------------------------------
> -------------------
> 28 24.646137 113.100.26.222 61.216.159.110 RTP
> 0.031826
> Arrival Time: Nov 23, 2009 23:50:32.660458000
> Sequence number: 7867
> Timestamp: 365000
>
>
> ------------------------------------------------------------------------
> --------------------
> 29 24.656106 113.100.26.222 61.216.159.110 RTP
> 0.009969
> Arrival Time: Nov 23, 2009 23:50:32.670427000
> Sequence number: 7868
> Timestamp: 365160
>
> ------------------------------------------------------------------------
> --------------------
>
> 30 24.675980 113.100.26.222 61.216.159.110 RTP
> 0.019874
> Arrival Time: Nov 23, 2009 23:50:32.690301000
> Sequence number: 3771
> Timestamp: 422060
>
> ------------------------------------------------------------------------
> ---------------------
> 31 24.685764 61.216.159.110 113.100.26.222 RTP
> 0.009784
> Arrival Time: Nov 23, 2009 23:50:32.700085000
> Sequence number: 3767
> Timestamp: 421420
>
>
> ------------------------------------------------------------------------
> ----------------------
> 32 24.695953 113.100.26.222 61.216.159.110 RTP
> 0.010189
> Arrival Time: Nov 23, 2009 23:50:32.710274000
> Sequence number: 7870
> Timestamp: 365480
>
>
> ------------------------------------------------------------------------
> -----------------------
> 33 24.704766 61.216.159.110 113.100.26.222 RTP
> 0.008813
> Arrival Time: Nov 23, 2009 23:50:32.719087000
> Sequence number: 3768
> Timestamp: 421580
>
>
> ------------------------------------------------------------------------
> -----------------------
>
> Please if you help me in telling that how can I calculated the
> Interarrival jitter in steps in that case. I shall be very thanksful to
> you.
>
> Regards,
>
>
>
>
>
> ________________________________
>
> Date: Thu, 26 Nov 2009 09:23:21 +0000
> From: martin.r.mathieson@xxxxxxxxxxxxxx
> To: wireshark-users@xxxxxxxxxxxxx
> Subject: Re: [Wireshark-users] End to End VoIP delay calculation
>
> There is the RTCP roundtrip network propagation delay. If the
> necessary reports are present and properly formatted, there will be an
> expert info item for any calculations that may be made. You will need to
> enable this calculation in the RTCP dissector preferences.
>
> Here is the extract from RFC 3550, section 6.4.1, that describes
> how the calculation should be done:
>
>
> delay since last SR (DLSR): 32 bits
> The delay, expressed in units of 1/65536 seconds, between
>
> receiving the last SR packet from source SSRC_n and
> sending this
> reception report block. If no SR packet has been received
> yet
> from SSRC_n, the DLSR field is set to zero.
>
> Let SSRC_r denote the receiver issuing this receiver
> report.
>
> Source SSRC_n can compute the round-trip propagation delay
> to
> SSRC_r by recording the time A when this reception report
> block is
> received. It calculates the total round-trip time A-LSR
> using the
>
> last SR timestamp (LSR) field, and then subtracting this
> field to
> leave the round-trip propagation delay as (A - LSR -
> DLSR). This
>
>
>
> Schulzrinne, et al. Standards Track
> [Page 40]
>
>
> RFC 3550 RTP
> July 2003
>
>
> is illustrated in Fig. 2. Times are shown in both a
> hexadecimal
> representation of the 32-bit fields and the equivalent
> floating-
>
> point decimal representation. Colons indicate a 32-bit
> field
> divided into a 16-bit integer part and 16-bit fraction
> part.
>
> This may be used as an approximate measure of distance to
> cluster
> receivers, although some links have very asymmetric
> delays.
>
>
> [10 Nov 1995 11:33:25.125 UTC] [10 Nov 1995 11:33:36.5
> UTC]
> n SR(n) A=b710:8000 (46864.500
> s)
>
> ---------------------------------------------------------------->
> v ^
>
> ntp_sec =0xb44db705 v ^ dlsr=0x0005:4000 (
> 5.250s)
> ntp_frac=0x20000000 v ^ lsr =0xb705:2000
> (46853.125s)
> (3024992005.125 s) v ^
> r v ^ RR(n)
>
>
> ---------------------------------------------------------------->
> |<-DLSR->|
> (5.250 s)
>
> A 0xb710:8000 (46864.500 s)
> DLSR -0x0005:4000 ( 5.250 s)
>
> LSR -0xb705:2000 (46853.125 s)
> -------------------------------
> delay 0x0006:2000 ( 6.125 s)
>
> Figure 2: Example for round-trip time computation
>
>
>
>
>
>
>
> On Thu, Nov 26, 2009 at 2:48 AM, Martin Visser
> <martinvisser99@xxxxxxxxx> wrote:
>
>
> As RTP in each direction is unacknowledged (you have a
> unidirectional stream going each direction) there is no way to determine
> end-to-delay from that. I think the best you can do is look at the SIP
> request/response time as an estimate.
>
> Regards, Martin
>
> MartinVisser99@xxxxxxxxx
>
>
>
> On Wed, Nov 25, 2009 at 4:31 AM, capricorn 80
> <cool_capricorn80@xxxxxxxxxxx> wrote:
>
>
>
> Hi!
>
>
> (Sorry for repeating my question)
>
> 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.
>
> Regards,
>
>
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