AW: [sip-comm-dev] How to affect the DELAY, PACKET LOSS and JITTER in SC


#1

Hello Emil, hello all,

I'v looked the packages in:

net.java.sip.communicator.impl.media.transform.*but I'm still not sure, which package is relevant for my work. You may be exacting.

You can do that with an RTP connector that would intercept all packets
before sending them. Our SRTP/ZRTP implementations do that so you can
have a look at their code for inspiration. Try starting here:

net.java.sip.communicator.impl.media.transform.TransformConnector

I don't understand. The RTP Connecor is a part from the JMF. That means im am not able to make

any changes (to manipulate the Packet Loss or Delay) of the source code in JMF. And SRTP/ZRTP
are by default in the SIP Communicator disabled, or when I use SIP Communicator and other device,
that not support SRTP or ZRTP, then I have a normal RTP connection. Is this right or I had
misunderstood something?

In your case you won't actually be modifying them but you could still
use the connector to delay and drop individual packets.

What you mean? You men the TransformConnector? And how is this possible without any modifying?

Thanks for your help

BR Nikolay

···

________________________________
Von: Emil Ivov <emcho@sip-communicator.org>
An: dev@sip-communicator.dev.java.net
Gesendet: Donnerstag, den 2. Juli 2009, 13:18:44 Uhr
Betreff: Re: AW: [sip-comm-dev] How to affect the DELAY, PACKET LOSS and JITTER in SC

I would like to produce an larger delay by introducing changes in the
source code of SC. That means for example - the voice packets will be
transmitted after 50, 100,500 ms or more from SC, not immediately. I
would like examining how the delay exactly affects the quality. I mean
the same with packet loss. I would like to modify the source code so,
that every 3, 5, 10, 15 data packet will be not send.

You can do that with an RTP connector that would intercept all packets
before sending them. Our SRTP/ZRTP implementations do that so you can
have a look at their code for inspiration. Try starting here:

net.java.sip.communicator.impl.media.transform.TransformConnector

In your case you won't actually be modifying them but you could still
use the connector to delay and drop individual packets.

....

Hope this helps!
Emil


#2

Hey Nikolay,

Nikolay Velichkov wrote:

Hello Emil, hello all,

I'v looked the packages in:

net.java.sip.communicator.impl.media.transform.*

but I'm still not sure, which package is relevant for my work. You may
be exacting.

I don't really know the specifics of your work but I'd guess that the
media package in general and transform in particular are those that
would be most affected.

You can do that with an RTP connector that would intercept all packets
before sending them. Our SRTP/ZRTP implementations do that so you can
have a look at their code for inspiration. Try starting here:

net.java.sip.communicator.impl.media.transform.TransformConnector

I don't understand. The RTP Connecor is a part from the JMF. That means
im am not able to make
any changes (to manipulate the Packet Loss or Delay) of the source code
in JMF.

You don't need to make changes to the code of JMF. That's the purpose of
the RTPConnector interface. It allows you to intercept and modify all
data before it gets to the network.

And SRTP/ZRTP
are by default in the SIP Communicator disabled,

Well they are actually on by default but this doesn't really matter in
your case. I only mentioned SRTP/ZRTP as code that you could use for
inspiration.

or when I use SIP
Communicator and other device,
that not support SRTP or ZRTP, then I have a normal RTP connection.

Yes. You'd need to:

1) modify the transform connector so that it stops doing SRTP and add
your modification code in it.
2) also modify CallSessionImpl and make sure that it always uses the the
connector and not an RTPManager.

In your case you won't actually be modifying them but you could still
use the connector to delay and drop individual packets.

What you mean? You men the TransformConnector? And how is this possible
without any modifying?

I meant that you won't be modifying packets. That is, you are not going
to change their content but only delay them or drop them. You can use
the transform connector to do this.

Cheers
Emil

···

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