I need to raise my ptime to 60 on my codecs for outbound calls. To that
effect, I add on the endpoint
disallow=all
allow=ulaw:60
and also
use_avpf : false
use_ptime : true
But the invites always leave with ptime:20.
It used to work fine in
To my knowledge, Asterisk's packetization rate is hard coded at 30ms. If I
wanted to, where in the code could I go to change it to 20ms. Is there
anything bad that might happen if I change it (asterisk related)?
___
--Bandwidth and Colocation
Matt wrote:
To my knowledge, Asterisk's packetization rate is hard
coded at 30ms. If I wanted to, where in the code could
I go to change it to 20ms. Is there anything bad that
might happen if I change it (asterisk related)?
You don't mention what version you are using, but 1.4 does
I am currently using 1.2 and can not upgrade to 1.4 until it becomes stable
and we have done much testing with it.
Obviously somewhere in the asterisk code 30ms must be coded... is it set in
just one place, and if so can I set that to 20ms?
On 3/14/07, Dan Austin [EMAIL PROTECTED] wrote:
Matt
Obviously somewhere in the asterisk code 30ms must be coded... is it set in
just one place, and if so can I set that to 20ms?
The default is 20 ms for most (all?) codecs. It's in rtp.c, where
ast_rtp_write() creates a new smoother.
--Luki
___
Oh? I thought the default was 30ms. Ok, if it's ALREADY at 20ms, then I
have nothing to modify.
On 3/14/07, Luki [EMAIL PROTECTED] wrote:
Obviously somewhere in the asterisk code 30ms must be coded... is it set
in
just one place, and if so can I set that to 20ms?
The default is 20 ms for
Vij wrote:
I have seen a few threads where people have applied packetization
patch and have varied the packetizing rate of RTP/SIP and hence
reducing the bandwidth required for the call.
Is there a way to do the same with IAX?.
Not at this time. There was some discussion about making the
Does this mean sending multiple frames per packet not possible with IAX currently?
Thanks,
VijOn 5/24/06, Dan Austin [EMAIL PROTECTED] wrote:
Vij wrote:I have seen a few threads where people have applied packetization patch and have varied the packetizing rate of RTP/SIP and hence reducing the
Vij wrote:
Does this mean sending multiple frames per packet not possible with
IAX
currently?
The answer to your question is tricky. Asterisk does send multiple
frames per packet now. The catch is that it is ALWAYS the number
of frames required for 20ms of audio (160 for 711, 20 for 729,
Hi,
I have seen a few threads where people have applied packetization
patch and have varied the packetizing rate of RTP/SIP and hence
reducing the bandwidth required for the call.
Is there a way to do the same with IAX?.
Will the tos=0x08 (highthroughput), or using the bandwidth directive
Is it possible in * to set the Packetization period. For example: If
I want G711 to be at
10ms. Is that possible in *?
Thanks,
Kurt
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Asterisk-Users mailing list
IAX is not an option as Sipura devices do not support AIX.Yes, the sipura will handle the different packet sizes...
Is it possible to reprogram asteris to do this?
On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote:On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware that
The packet size is a function of the number of milliseconds of sound sent
in the RTP packet. I don't know how to force * to change this, but you
*can* unilaterally change the RTP packet size on the Sipura. By doing
this, RTP packets sent by the Sipura will be larger or smaller than the
default
Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok... where do you change it on the sipura?On Apr 3, 2005 4:07 PM, Bruce Komito [EMAIL PROTECTED] wrote:The packet size is a
Never mind... blah spoke before I thought :P
Found the setting
On Apr 3, 2005 5:23 PM, Matt [EMAIL PROTECTED] wrote:Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok...
I have to admit this still doesn't make sence.. if sipura's default is
.03ms and asterisk is 20ms.. why is the sipura dumping out around 60
frames/sec while the sipura is dumping out around 30 frames/sec??
Shouldn't the frames / packets per second go UP as the packetization gets smaller?On Apr 3,
I'm aware that asterisk only supports 20ms packetization rates.
Due to the fact that I will be using some voip devices on a wireless
network which is highly sensative to framerate.. is there any way I can
hard code the packetization rate at say 30 or 40ms and then compile
astrisk? If so, can
On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:
I'm aware that asterisk only supports 20ms packetization rates. Due
to the fact that I will be using some voip devices on a wireless
network which is highly sensative to framerate.. is there any way I
can hard code the packetization rate at say
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