Wireshark mailing list archives

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


From: "RUOFF LARS" <Lars.Ruoff () alcatel-lucent com>
Date: Fri, 27 Nov 2009 16:11:29 +0100

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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