Re: problem while reducing the seding rate

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!
>>>
>>
> 
>