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Re: End to End VoIP delay calculation (Interarrival jitter)

From: capricorn 80 <cool_capricorn80 () hotmail com>
Date: Sun, 27 Dec 2009 15:50:44 +0000

 

Hi Martin !

 

 Sorry i was out of town and didnt have access to my emails. I will post my question in clear form. 

 

Regards,


 


Date: Mon, 30 Nov 2009 13:27:06 +1100
From: martinvisser99 () gmail com
To: wireshark-users () wireshark org
Subject: Re: [Wireshark-users] End to End VoIP delay calculation (Interarrival jitter)

Why is *what* the case? Your question isn't clear.


If you want to see RTP statistics on your stream do the following


1. Select an RTP packet
2. Go to the Telephony menu and select RTP -> Stream Analysis.Regards, Martin

MartinVisser99 () gmail com



On Sat, Nov 28, 2009 at 10:17 AM, capricorn 80 <cool_capricorn80 () hotmail com> wrote:




 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 () alcatel-lucent com



To: wireshark-users () wireshark org
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 () wireshark org
[mailto:wireshark-users-bounces () wireshark org] On Behalf Of capricorn 80
Sent: vendredi 27 novembre 2009 15:44
To: wireshark-users () wireshark org
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 () googlemail com
To: wireshark-users () wireshark org
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 () gmail com> 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 () gmail com



On Wed, Nov 25, 2009 at 4:31 AM, capricorn 80
<cool_capricorn80 () hotmail com> 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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