[Freeswitch-users] frame per packet

[David Ponzone]

Kenneth,

It seemed to me that it's the endpoint which must first decide the ptime it wants to use.
Then FreeSWITCH will relay it.
What you did will probably help FreeSWITCH to advertise only GSM with 60ms ptime to the other endpoint, but I think you may have achieved that by using:
GSM at 60i
in your codec strings.

David Ponzone  Direction Technique
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Le 22/04/2011 à 07:56, Kenneth Taylor a écrit :

> Hello,
> I think I found a solution.
> At mod_spandsp_codecs_load function on mod_spandsp_codecs, there is a place where FS is loading all of its codecs.
> for the gsm codec (the codec i'm using) it is loading 6 implementations of the codec sorted by the number of frames per network packet.
> When a conversation is starting FS choose the last implementation which is for 1 frame per packet.
> I disabled all the implementations except the one with 3 frames per packet, and now it's working.
> 
> Just to make sure I'm not missing anything.
> 
> TNX everyone
> Ken
> 
> 
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Mon, April 18, 2011 11:04:40 PM
> Subject: Re: [Freeswitch-users] frame per packet
> 
> AFAIK, a rtp packet *is* a frame, so no you can't.
> 
> As the other poster said, use a higher ptime. That will make the frame store a longer period of time which will do the same as that you want.
> 
> Overhead will be lower, but quality will be worse if you drop packets or they fail to arrive in a timely manner. That's very likely on gsm, and the packet size will mean it takes longer to be delivered which'll mean more delay and probably more packets arriving too late. You'll need to experiment to find a good balance. And remember what works well in a city might fail to work in the country where coverage is poorer.
> 
> Steve on iPhone
> 
> On 18 Apr 2011, at 09:49, Kenneth Taylor <ktaylor91 at yahoo.com> wrote:
> 
>> Hi,
>> I'm building a voip client for android phones over gsm and consider using FS as sever.
>> In order to save cellular costs I want to put more than one frame per Rtp packet.
>> 
>> Is there any easy solution to do it in FS? 
>> 
>> Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go to lower levels?
>> I don't think I should do it in the codec, it may cause problems in the rtp sequence number and probably a lot more.
>> 
>> TNX,
>> Ken
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