Sorry, I didn't mean the variable bit rate codecs. My doubt is
regarding any kind of data (audio, video etc).
Suppose we are using something like below:
then, the program sends only 3 packets. Whereas, if the above
statement is changed to
the program sends some 30+ packets.
Now, when I am writing a program to send data (be it of any
kind), I may use the program to send 1 MB of data or even
50 MB of data; so to what value should the expire timeout
be set ?
I hope, I have made myself clear this time around :)
I assume the value given to setExpireTimeout is in microseconds;
else please correct me.
On Thu, 12 May 2005, Michel de Boer wrote:
Why in the first place do you want to transmit the data at
a different speed? What codec do you use for the audio?
I have only experience with G.711 and GSM encoded audiostreams.
These codecs have a constant bit rate, eg. G.711 has 160 bytes/20ms,
GSM has 33 bytes/20ms So I have to make sure to send the audio streams
with these speeds. I just set the correct payload type with
setPayloadFormat and it all works fine as long as I deliver the
data at the correct rate to ccRTP.
I am not sure how RTP works with variable bit rate codecs.
Dinil Divakaran wrote:
For the time being I fixed sending 2833 events as follows:
rtp_session->setExpireTimeout(duration of event)
This way the packets stay within the oldness check in ccRTP.
But, we can not give a static argument to setExpireTimeout since
the number of packets change depending on the data that has to
be transmitted. Hence, if the value set by setExpireTimeout is
okay for a 1 MB data, it need not be useful for sending data
larger than 1 MB. This happens since the packets do not stay
within the oldness check as the number of packets increase.
David Sugar wrote:
Oh, this is about sending 2833 events, not receiving...sorry :).
Hmm, let me think about this one further!