AW: [sip-comm-dev] How to affect the DELAY, PACKET LOSS and JITTER in SC
emcho at sip-communicator.org
Wed Jul 15 14:46:36 CEST 2009
Nikolay Velichkov wrote:
> Hello Emil, hello all,
> I'v looked the packages in:
> 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:
> 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
> 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.
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