, 2003 10:20 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] frames/packet
Hi,
I used Cisco 3640 with 2xNM-HDV-2E1 cards.
The default GW router has RTP and TCP/UDP header compressions. There is
also a Linux solution for this. You can run RTP compression on your
asterisk box, and or run UDP
the moment ?
Cheers,
Abdul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Luyens
Sent: 29 September 2003 15:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] frames/packet
Hi Abdul, can you tell which hardware (CPU, Mem) you used to manage the
PROTECTED]
Subject: RE: [Asterisk-Users] frames/packet
Hi,
A bit late replying to this.
My comments are below:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Lambert
Sent: 03 September 2003 17:16
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users
Hi,
A bit late replying to this.
My comments are below:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Lambert
Sent: 03 September 2003 17:16
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] frames/packet
"Not yet." implies that it is
t; <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, September 03, 2003 5:15 PM
Subject: Re: [Asterisk-Users] frames/packet
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Don't count on being able to push the full 32 simultaneous calls. Cable
modems are notorious for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Don't count on being able to push the full 32 simultaneous calls. Cable
modems are notorious for failing to deliver the full bandwidth the subscriber
is paying for. In many places customers end up getting 33Kbbps or less on a
regular basis. The a
We have customers all over the world on our SIP Services (not Asterisk Based).
We have found the optimal setting to be 60ms on the phones (using G729).
This consumes about 14kbps of bandwith per call. The added delay is quite
acceptable and customers often tell us that the voice quality is mu
I'm thinking about SIP as well.
Martin Pycko wrote:
>
> Unless you use IAX2 trunking that would limit 32 calls to 50 packets/sec.
>
> Martin
>
> On Wed, 3 Sep 2003, Paul Lambert wrote:
>
> > "Not yet." implies that it is coming. I know it would help on Internet
> > connections such as fixed wi
Steve Underwood wrote:
>
> Paul Lambert wrote:
>
> >"Not yet." implies that it is coming.
> >
> Look at the latency it causes, and you will see its not that useful.
>
> >I know it would help on Internet
> >connections such as fixed wireless and cable modem where packet rate is
> >an issue. 20ms
That is not just true of IAX. There appears to be substantial amount of
RTP traffic, which trunks a variable bundle of calls between the same
two points, used by IDD services. The traffic has to be going between
the same two points to make that work, though, whichever protocol you
use as the tr
Paul Lambert wrote:
"Not yet." implies that it is coming.
Look at the latency it causes, and you will see its not that useful.
I know it would help on Internet
connections such as fixed wireless and cable modem where packet rate is
an issue. 20ms translates to 50 packets/sec.
30ms per block c
Unless you use IAX2 trunking that would limit 32 calls to 50 packets/sec.
Martin
On Wed, 3 Sep 2003, Paul Lambert wrote:
> "Not yet." implies that it is coming. I know it would help on Internet
> connections such as fixed wireless and cable modem where packet rate is
> an issue. 20ms translates
"Not yet." implies that it is coming. I know it would help on Internet
connections such as fixed wireless and cable modem where packet rate is
an issue. 20ms translates to 50 packets/sec. I believe cable modem
upstream packet rates cap at 150-200 packets/sec. G729 gets the bit rate
down to 8kbits.
Not yet.
Asterisk always sends 20 ms of voice data per packet.
regards
Martin
On Wed, 3 Sep 2003, Paul Lambert wrote:
> Noticed that I can adjust the number if frames/packet on the GrandStream
> phone. Can * do the same?
> ___
> Asterisk-Users mailin
14 matches
Mail list logo
