[Asterisk-Dev] Identifying associated channel in rtp.c

[Alf Thomas Nilsen]

I am trying to make it possible to define how much voice data one would like
to transmit with each packet for certain codecs and protocols. This is meant
to be used where bandwidth is very expensive and hopefully it could reduce
some overhead. I am aware of potential problems but I will try to test and
weigh cost vs. quality against each other. It might not be a good idea, but
I'd really like to test it :)

I am hoping to be able to define the amount of voice data in each packet
through extensions.conf. That way several conversations at the same time
will be able to operate at different voice payloads.

Best regards,
Alf Thomas Nilsen

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Brian West
Sent: 19. april 2005 20:48
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] Identifying associated channel in rtp.c

what is your goal?  Maybe if you tell us that we could recommend 
something better.

/b

On Apr 19, 2005, at 9:38 AM, Alf Thomas Nilsen wrote:

> So if I'd like to do a quick and dirty modification I could modify the
> ast_rtp_write() method to take a third parameter (the retrieved 
> variable),
> and retrieve the variable from within sip_write() by calling:
>
> myvar = atoi(pbx_builtin_getvar_helper(p->owner,"myvar"));
>
> I would then pass the variable as a parameter in the ast_rtp_write() 
> call
> inside sip_write(). The variable should then be available from within
> ast_rtp_write() in rtp.c.
>
> I do, however, acknowledge that this is a really "ugly" modification, 
> and
> I'm somewhat shameful to propose such a "solution" to this problem.
>
> Best regards,
> Alf Thomas Nilsen
>
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Kevin P. 
> Fleming
> Sent: 19. april 2005 15:32
> To: Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] Identifying associated channel in rtp.c
>
> If you want to do this, you'll need to retrieve the channel variable in
> chan_sip (or whatever RTP-using channel you are working with), then 
> pass
> the value into the RTP layer via some other means.
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