Wireshark mailing list archives
Re: End to End VoIP delay calculation (Interarrival jitter)
From: capricorn 80 <cool_capricorn80 () hotmail com>
Date: Fri, 26 Feb 2010 09:32:55 +0000
Hi! Thanks for your response. Is it possible to come up with some results or conclusion with this data? Regards, From: gpeaslee () verizon net To: wireshark-users () wireshark org Date: Sat, 16 Jan 2010 12:24:54 -0600 Subject: Re: [Wireshark-users] End to End VoIP delay calculation (Interarrival jitter) I'd guess that what you're seeing in the laptop trace is two RTP streams (normal). If you look at the SSRC in the RTP packet you'll see they're different based on what the source address is. the aaa.pcap looks like it's only showing one of the streams. End to end jitter is a meaningless calculation, jitter only effects one stream at a time and RTP isn't really a round trip event, the speech is two one way streams. ----- Original Message ----- From: capricorn 80 To: wireshark-users () wireshark org Sent: Saturday, January 16, 2010 11:05 AM Subject: Re: [Wireshark-users] End to End VoIP delay calculation (Interarrival jitter) Hi ! I will try to explain my question. This is the output from aaa.pcap file. If you see in all cases the source address is 192.168.1.2 and destination address is 212.242.33.36. No Time Source Destination Protocol Delta time 624 1444.509099 192.168.1.2 212.242.33.36 RTP 0.113797 625 1444.579046 192.168.1.2 212.242.33.36 RTP 0.069947 626 1444.582579 192.168.1.2 212.242.33.36 RTP 0.003533 627 1444.588245 192.168.1.2 212.242.33.36 RTP 0.005666 628 1444.590352 192.168.1.2 212.242.33.36 RTP 0.002107 This is my reading from my laptop where source 213.100.26.x is my laptop and destination is asterisk server with IP adress 81.216.x.x No Time Source Destination Protocol Delta time 28 24.646137 213.100.26.x 81.216.x.x RTP 0.031826 29 24.656106 213.100.26.x 81.216.x.x RTP 0.009969 30 24.675980 213.100.26.x 81.216.x.x RTP 0.019874 31 24.685764 81.216.x.x 213.100.26.x RTP 0.009784 32 24.695953 213.100.26.x 81.216.x.x RTP 0.010189 33 24.704766 81.216.x.x 213.100.26.x RTP 0.008813 My question was that in the example aaa.pcap, 192.168.1.2 is always the source and 212.242.33.36 is always the destination but if you see the output from my laptop sometimes the source is 213.100.26.x and some times its 81.216.x.x. why in my case the IP address 213.100.26.x is not always the source ??? Thanks for your help. From: cool_capricorn80 () hotmail com To: wireshark-users () wireshark org Date: Sun, 27 Dec 2009 15:50:44 +0000 Subject: Re: [Wireshark-users] End to End VoIP delay calculation (Interarrival jitter) 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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Current thread:
- Re: End to End VoIP delay calculation (Interarrival jitter) capricorn 80 (Feb 26)