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:

    ------------------------------------------------------------------------

    --------------------

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:

    ------------------------------------------------------------------------

    ---------------------

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:

    ------------------------------------------------------------------------

    -----------------------

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:

    ------------------------------------------------------------------------

    -----------------------


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

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

<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)