[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
More information about the asterisk-users
mailing list