Wireshark mailing list archives
Re: End to End VoIP delay calculation (Interarrival jitter)
From: capricorn 80 <cool_capricorn80 () hotmail com>
Date: Fri, 27 Nov 2009 23:17:10 +0000
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. 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Current thread:
- End to End VoIP delay calculation capricorn 80 (Nov 22)
- vierify_tools problem in 1.3.1 Reddy Nagendra-GKTC37 (Nov 22)
- <Possible follow-ups>
- End to End VoIP delay calculation capricorn 80 (Nov 24)
- Re: End to End VoIP delay calculation Martin Visser (Nov 25)
- Re: End to End VoIP delay calculation Martin Mathieson (Nov 26)
- Re: End to End VoIP delay calculation (Interarrival jitter) capricorn 80 (Nov 27)
- Re: End to End VoIP delay calculation (Interarrival jitter) RUOFF LARS (Nov 27)
- Re: End to End VoIP delay calculation (Interarrival jitter) capricorn 80 (Nov 27)
- Re: End to End VoIP delay calculation (Interarrival jitter) Martin Visser (Nov 29)
- Re: End to End VoIP delay calculation Martin Visser (Nov 25)