[Iaxclient-devel] problem with IAX2 timestamps (and newbe introduction...)

[<an...@mo...>]

Hi,

First, let me introduce myself... I'm Moreiras, from Brazil, new to this
list, and new to the "asterisk" word. I'm working with asterisk for two
months and with iaxclient and iaxcomm (in Windows) for a little more than
1 month.  I think I'm a newbe in this hole thing.

In time, I think my english is not very good. But I will try to be clear...

I have three questions about the library, that I'm going to ask in
separated emails.

First one: I have sound problems in this configuration:

IAXCOMM Windows (GSM) ---> Asterisk Server ---> Cisco SIP IP Phone (ULAW)

The sound quality in IAXCOMM side is OK. But at the Cisco side, it is
horrible. Looking in asterisk email lists, bugs, and google I found that
probably it is a problem with generated IAX2 timestamps. The problems is
already resolved in asterisk server. But maybe it is not in Iaxcomm /
iaxclient library, I don't know for sure.

The timestamps generated by iaxcomm are a little erratic. There are
increments between 15ms and 30ms between the timestamps of consecutive
packets. (One can see that with tcpdump, for example). The asterisk server
gets that timestamps then generate the SIP timestamps... And the CISCO
phone doesn't like erratic timestamps, it drop some packets, and the sound
quality go down...

The following configuration sounds OK:

IAXCOMM (GSM)  ---->  Asterisk Server Meeetme app <---- Cisco SIP (ULAW)

I think that in this configuration the server generate the timestamps
without look at that generated by iaxcomm...  everything works fine.

At iax.c, in the library (libiax2), I see that there is some code to
generate the timestamps based on the timing from the audio sampling
(8khz). The code is not used by default, the variable
USE_VOICE_TS_PREDICTION have to be defined to do this.

Well, I put a:
#define USE_VOICE_TS_PREDICTION 1
at the beginning  of iax.c.

Everything works fine, then. Time difference between timestamps is
something more than 20ms, near 23ms, but it is ever 23ms, no more erratic
differences, and it seams to don't generate sound problems. And the sound
at the SIP ULAW CISCO side is OK, now.

Well...

I'd like to know if there is a problem that I don't see with this code
(VOICE_TS PREDICTION). Maybe at other plataforms (I'm using windows...)?
If no, why this is not the default choice?

At asterisk lists I see that this problem (erratic timestamps with IAX2
and GSM) was resolved in asterisk, but I was not able to identify this
correction looking at the code (not so good programmer... hehehe) Is there
some other way to do this?

Thanks!

Moreiras.







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