Wireshark mailing list archives
Re: End to End VoIP delay calculation (Interarrival jitter)
From: "George Peaslee" <gpeaslee () verizon net>
Date: Sat, 16 Jan 2010 12:24:54 -0600
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 (Jan 16)
- Re: End to End VoIP delay calculation (Interarrival jitter) George Peaslee (Jan 16)
- temp files not cleaned up Jeff Liegel (Jan 16)
- Re: temp files not cleaned up Jeff Morriss (Jan 18)
- temp files not cleaned up Jeff Liegel (Jan 16)
- Re: End to End VoIP delay calculation (Interarrival jitter) George Peaslee (Jan 16)