[Asterisk-Users] Asterisk codecs and packet size

Andres andres at telesip.net
Thu Sep 2 17:04:31 MST 2004


Michael Manousos wrote:

> Andres wrote:
>
>>
>>>
>>> The quick and dirty way:
>>> ------------------------
>>>
>>> In rtp.c, function "ast_rtp_write", in the "switch" statement,
>>> "AST_FORMAT_G729A" case, change the smoother creation to something
>>> larger. E.g.:
>>>
>>>     rtp->smoother = ast_smoother_new(40);
>>>
>>> Keep in mind that you must set this into something valid
>>> (45 obviously is not valid). Recompile and you should be fine.
>>>
>> Michael, this little nugget made my day.  Last year we offered to pay 
>> for this development.  Too bad you didn't collect:)
>
>
> Just out of curiosity. What was the offering for this one-line
> patch?

We simply sent an email to the list asking all interested developers to 
give us a quote.  We also asked Digium to give us a quote.  Nobody even 
replied or showed any interest.  From this we deducted this development 
effort was so complex nobody wanted to take a stab at it.  Last year we 
had budgeted US$1000 for this effort.  We were also trying to recruit 
others interested in this to chip in some money.  Nobody answered 
either.  We had given up on this a long time ago...and the budget was 
spent on other things.

>
>>
>> Thanks!
>>
-- 
Andres
Network Admin
http://www.telesip.net





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