Ethereal-users: Re: [Ethereal-users] RTCP for VoIP QoS
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I dont mean to discourage u. But u wouldnot find any currently. I personally suggest (if u r so motivated) to comile the openH323 on ur machine and try to debug it. Compiling the stack could be a bit of problem. But then u could find some support on the OpenH323 project's mailing list. Then if u r a reasonable coder, u can debug the code (or activate the fields). But its all up to u.
Regards,
Adil Raja
Hi Adil,
Thank you for the reply. Yes unfortunately I am after finding that XLite send bogus RTCP values and don't seem to implement the RTP/RTCP stack correctly. I looked at the Counterpath Xlite mailing list and found other people who have had similar issues. This is very disappointing so I will now need to find a softphone which sends correct RTCP information.
Regards,
Aisling.
-----Original Message-----
From: ethereal-users-bounces@xxxxxxxxxxxx [mailto:ethereal-users-bounces@xxxxxxxxxxxx] On Behalf Of Adil RajaSent: 28 March 2006 13:04
To: Ethereal user support
Subject: Re: [Ethereal-users] RTCP for VoIP QoS
David,
Well thats what I meant to say.
Regards,
Adil RajaOn 3/27/06, David Grau Serra < dgs@xxxxxxxxxx > wrote:
Hi Adil,
SIP is a signalling protocol.
RTP is a transport of real-time data protocol.
Media path != signalling path.
Every RTP session has a parallel RTCP session.
RTCP:
* Exchange information about losses and delays between the end systems.
* Packets sent in intervals determined based on number of end
systems and available bandwidth.
* The sender knows what quality of reception the receiver is
experiencing.
I just know that some extensions to RTCP are called RTCP-XR.
Regards,
David
> David,
> It is good to hear from u. The best thing I worked was the OpenH323
> project. It does not have some calculations implemented properly (such
> as LSR or DLSR). I try to probe into the code but finding that it was
> quite a bit of work and that a deviation from my work I left it for
> some solitary time. I have also tried with winRtp as well but it had
> some problems as well. Astonishingly, the stach has some bugs in some
> of its basic calculations. Anyway,
> The thing with SIP is that it is call control and signalling protocol
> (a suite so to say). RTCP always comes as a part of RFC 3550. i.e. the
> RTP protocol.
>
> I must share with u one more thing (or if I am lagging in knowledge
> then please correct me) that RTCP-XR is not supported by any of the
> applications I have encountered so far. This means that even if there
> are some they are only a few. Which implies that the number of ppl
> using such applications are also few. On the other hand some companies
> are providing solutions to masses based on the assumption that RTCP-XR
> is there. Anyway, it was just a thought and I wanted to share it with u.
>
> Regards,
> Adil Raja
>
> On 3/27/06, *David Grau Serra* <dgs@xxxxxxxxxx
> <mailto:dgs@xxxxxxxxxx>> wrote:
>
> Hi Adil,
>
> I am helping Aisling, and if I understand what are you saying,
> these it
> could happen to xlite phone, but I am not sure at all.
> I give you some links that they are saying that:
>
> http://support.counterpath.net/viewtopic.php?t=2154
> http://support.counterpath.net/viewtopic.php?t=2201
> http://support.counterpath.net/viewtopic.php?t=5920
>
> By the way, I can catch RTCP packets in both directions using xlite
> phone, but as you say, maybe the specific fields for delay
> calculations
> are not implemented properly...
> I am wondering, like you, if there are any soft phone (SIP) hich
> handles
> this effectively.
>
> > Aisling,
> > I would just like to add a few things here. I have seen a
> number of
> > implementations of RTP and in all the cases either there are no
> RTCPs
> > or if they are present then the specific fields for delay
> calculations
> > are not implemented properly (such as in the ohphone stack). So, I
> > just wanted to say this. To the contrary, if u find an
> implementation
> > which handles this effectively then please let me know.
> >
> > Regards,
> > Adil Raja
> >
> > On 3/27/06, *Martin Mathieson* <martin.mathieson@xxxxxxxxxxxx
> <mailto:martin.mathieson@xxxxxxxxxxxx>
> > <mailto:martin.mathieson@xxxxxxxxxxxx
> <mailto: martin.mathieson@xxxxxxxxxxxx>>> wrote:
> >
> > Aisling, I've written some answers below prefixed by [Martin].
> >
> > Hello Martin,
> >
> > My name is Aisling and I am helping David Grau with his
> project (his
> > project as part of an undergraduate degree but this also
> interests me
> > as part of my postgraduate degree).
> >
> > I have read rfc 3550 with particular emphasis on the part
> detailing
> > how the round trip time (RTT) should be calculated. The formula
> > "A-DLSR-LSR" is provided and an example " 46864.500 - 5.250 -
> > 46853.125 = 6.125 seconds" is given. We are working from a
> tethereal
> > capture file - This was running on the same pc as one of the
> > softphones involved in the call but as the two softphones
> calling
> > each other are on the same LAN segment we should have
> captured all
> > the necessary RTCP sender and receiver reports anyway.I have
> attached
> > the sample capture file.
> >
> > [Martin] Note that within the same LAN segment you are
> unlikely
> > to see any
> > interesting roundtrip delays. You can set the RTCP preference
> > 'Minimum rountrip calculations to report'
> > to 0 milliseconds.
> >
> >
> >
> > Myself and David are not clear on a few things:
> >
> > 1) The A field is when the receiver report block is
> received. Is this
> > referring to the NTP timestamp? I am struggling to identify
> this from
> > the tethereal capture (perhaps the capture format is slightly
> > non-standard?)and if you could point this out from the attached
> > capture file we would greatly appreciate it.
> >
> > [Martin] 'A' seems to be the middle 32 bits from the NTP
> > timestamp, whose units would be 1/65536th seconds.
> > All 3 quantities in the formula you quote are in these units,
> > though they are converted into seconds to make it easier to
> read.
> >
> >
> >
> > 2)The LSR field is said to be the middle 32 bits of the NTP
> timestamp
> > and the NTP timestamp is made up of the MSW and LSW. So if the
> > figures are MSW=3350115113 and LSW=493921239 I still fail to
> > understand why the LSR=3005816176...this is based on the
> figures from
> > the attach capture file.
> >
> > [Martin] MSW and LSW are being displayed as decimals, it is
> > easier to look at them as hex.
> > MSW = 0xC7AEB329, LSW = 0x0x1D70A3D7
> >
> > The middle word in this case from this would be 0xB3291D70 =
> > 3005816176
> >
> > However, remember that the LSR in a frame refers to the
> timestamp
> > seen in a message seen in the opposite direction
> > (i.e. in a received message it refers to the middle-bytes
> from the
> > time of a frame you previously sent).
> >
> >
> >
> > 3)the DLSR does appear to be 1 for all captures but as you
> said this
> > could be for the reasons that you described before.
> >
> > [Martin] As in the other example I looked at, there appears
> to be
> > only RTP/RTCP in one direction, so these values looke made up /
> > default to me.
> > You maybe need to make sure that someone is speaking in both
> > directions to that RTP and SRs are sent. No calculation can be
> > done until a SR in one direction
> > refers to an SR in the previous direction (by matching its
> LSR and
> > having a sensible DLSR filled in).
> >
> >
> >
> > 4)Based on the above points and the attached capture file
> could you
> > give an example with figures (from the file) of using
> A-DLSR-LSR?
> >
> > [Martin] Sorry, I don't have ready access to my old capture
> > collection at the moment. I realise that the way this
> calculation
> > is done is a bit complex.
> > I notice that RFC 3611 allows non-senders to calculate roundtrip
> > delays, but I've never seen it used in a real client.
> >
> > I really do hope this helps you.
> > Best Regards,
> > Martin
> >
> >
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