[Speex-dev] Speex questions

[José Miguel Galea Yrausquin]

Hi Jean-Marc, I'm implementing a VoIP client and I want to use your
codec init. I need to
ask you some questions about Speex because I can't get too much information
about it on the web. I hope you can help me.

1.- First of all, I want to know if Speex provides a buffer for Jitter..If
it does, what can I do to use it in my softphone?

2.- How does Speex manage the packet loss? (In a general point of view)

3.- Is the Delay for this codec low or high? I saw something strange in my
implementation. I was using constant bit rate, when I set VAD as enabled the
Delay was bigger. You told me by IRC that VAD doesn't have to do with the
Delay, but it is happening to me. What could it be?

4.- Another thing you told me by IRC is to pack two frames in one packet to
reduce the bandwidth consumption. I was packing just one frame per packet
and it worked. I got audio in both clients but the bandwidth consumption
wasn't as low as I wanted. Actually my softphone is consuming 3.5 KBytes/sec
(Speex+ RTPheader). So, I try to pack two frames per packet but the audio is
too bad. I call the speex_encode_int method before call the speex_bits_write
function. What could be happening with this?

5.- Finally, I want to know if it's possible to change the bandwidth
consumption on the fly. I mean, If I use the re-INVITE to change the
parameters of a session, is it possible to change the parameters of the
codec and then change the bandwidth consumption too?

I'm using QUALITY = 4 and COMPLEXITY = 3.

Hope you can help me. Is very important for me to use Speex in my softphone.
Its features are really good, but I can't use all of them because of the
troubles I have at the moment.

Thanks in advance.... José Miguel.
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