> I have a question regarding the Asterisk Packet Time for SIP Calls. It is
> hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that
> these packets are not spaced out at 20ms. In general you see something like:
>
> Packet 50 - Delay 50ms
> Packet 51 - Delay 5ms
> Packet
> > Packet 50 - Delay 50ms
> > Packet 51 - Delay 5ms
> > Packet 52 - Delay 5ms
> > Packet 53 - Delay 50ms
> > Packet 54 - Delay 5ms
> > Packet 55 - Delay 5ms
> The 20 ms is not the inter-packet timing, its the relative content of
> what's within the packet. In other words, the packet contains 20ms
ew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Monday, December 22, 2003 3:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
> > Packet 50 - Delay 50ms
> > Packet 51 - Delay 5ms
> > Packet 52 - Delay 5ms
> > Packet 53 - Delay 50ms
>
> > > Packet 50 - Delay 50ms
> > > Packet 51 - Delay 5ms
> > > Packet 52 - Delay 5ms
> > > Packet 53 - Delay 50ms
> > > Packet 54 - Delay 5ms
> > > Packet 55 - Delay 5ms
>
> > The 20 ms is not the inter-packet timing, its the relative content of
> > what's within the packet. In other words, the pa
On Monday 22 December 2003 15:36, Rich Adamson wrote:
> > I have a question regarding the Asterisk Packet Time for SIP Calls. It
> > is hardcoded at 20ms but when I do an RTP Analysis on a stream it is
> > clear that these packets are not spaced out at 20ms. In general you see
> > something like:
On Monday 22 December 2003 15:55, Andrew Kohlsmith wrote:
> > > Packet 50 - Delay 50ms
> > > Packet 51 - Delay 5ms
> > > Packet 52 - Delay 5ms
> > > Packet 53 - Delay 50ms
> > > Packet 54 - Delay 5ms
> > > Packet 55 - Delay 5ms
> >
> > The 20 ms is not the inter-packet timing, its the relative cont
On Monday 22 December 2003 16:37, Andres wrote:
> On Monday 22 December 2003 15:36, Rich Adamson wrote:
> > > I have a question regarding the Asterisk Packet Time for SIP Calls. It
> > > is hardcoded at 20ms but when I do an RTP Analysis on a stream it is
> > > clear that these packets are not spa
> On Monday 22 December 2003 16:37, Andres wrote:
> > On Monday 22 December 2003 15:36, Rich Adamson wrote:
> > > > I have a question regarding the Asterisk Packet Time for SIP Calls. It
> > > > is hardcoded at 20ms but when I do an RTP Analysis on a stream it is
> > > > clear that these packets a
On Monday 22 December 2003 19:58, Rich Adamson wrote:
> > On Monday 22 December 2003 16:37, Andres wrote:
> > > On Monday 22 December 2003 15:36, Rich Adamson wrote:
> > > > > I have a question regarding the Asterisk Packet Time for SIP Calls.
> > > > > It is hardcoded at 20ms but when I do an RTP
Rich Adamson wrote:
I have a question regarding the Asterisk Packet Time for SIP Calls. It is
hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that
these packets are not spaced out at 20ms. In general you see something like:
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
P
> >>Packet 50 - Delay 50ms
> >>Packet 51 - Delay 5ms
> >>Packet 52 - Delay 5ms
> >>Packet 53 - Delay 50ms
> >>Packet 54 - Delay 5ms
> >>Packet 55 - Delay 5ms
> >>
> >>Is there anyway to space them out evenly at 20ms??
> >
> >
> > The 20 ms is not the inter-packet timing, its the relative content
Interesting. For the record, the MultiTech MVP-130 comes with a default
setting
of 60ms packets on all of its supported codecs. I changed the packet
sizes to
20ms because I had never heard of anyone using such large sample sizes.
Andres wrote:
On Monday 22 December 2003 19:58, Rich Adamson wr
Olle,
Here is an interesting site that goes into some of the troubleshooting
techniques in Voip:
http://www.voiptroubleshooter.com/
Maybe it will help your FAQ!
Olle E. Johansson wrote:
Rich Adamson wrote:
I have a question regarding the Asterisk Packet Time for SIP Calls.
It is hardcoded at
On Tue, 23 Dec 2003, Rich Adamson wrote:
> If a collision or dropped packet occurs (in a voip udp environment) there
> is no way to retransmit the missing/damaged packet. Missing one packet isn't
> a big deal, but if you have collisions and/or dropped packets, there is a
> very high probability th
notice
it!
-Original Message-
From: Joel Maslak
To: [EMAIL PROTECTED]
Sent: 12/23/2003 10:41 AM
Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Tue, 23 Dec 2003, Rich Adamson wrote:
> If a collision or dropped packet occurs (in a voip udp environment)
there
&
I'm not sure under what circumstances (from an overall performance
perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
suggest choosing the size is roughly equivalent to MTU size. The 60ms
setting should result in larger packets which might be okay for high
speed uncongested l
application cannot get reassembled in a timely manner, you'll surely notice
> it!
>
> -Original Message-
> From: Joel Maslak
> To: [EMAIL PROTECTED]
> Sent: 12/23/2003 10:41 AM
> Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
>
> On Tue, 23
On Tuesday 23 December 2003 10:59, Rich Adamson wrote:
> I'm not sure under what circumstances (from an overall performance
> perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
In our network we set UAs to use 60ms (using G729). Actual data measurements
indicate a call consu
en a voice
> > application cannot get reassembled in a timely manner, you'll surely
> > notice it!
> >
> > -Original Message-
> > From: Joel Maslak
> > To: [EMAIL PROTECTED]
> > Sent: 12/23/2003 10:41 AM
> > Subject: Re: [Asterisk-Users]
semble it in a timely manner. It's
> not a
> > > big deal with a web page or something along that lines. But when
> a voice
> > > application cannot get reassembled in a timely manner, you'll
> surely
> > > notice it!
> > >
> > > -Original Me
> > > > application cannot get reassembled in a timely manner, you'll
> > surely
> > > > notice it!
> > > >
> > > > -Original Message-
> > > > From: Joel Maslak
> > > > To: [EMAIL PROTECTED]
> > > &g
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