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

["RUOFF LARS" <Lars.Ruoff@xxxxxxxxxxxxxxxxxx>]

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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