On Apr 30, 2006, at 9:03 AM, Eric ManxPower Wieling wrote:
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is
used to transport audio for SIP (and other protocols). This means
that ANY jitter on the SIP Phone - Asterisk link will cause audio
Eric ManxPower Wieling wrote:
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to
transport audio for SIP (and other protocols). This means that ANY
jitter on the SIP Phone - Asterisk link will cause audio problems.
This is only an issue if
This is only an issue if your SIP phone has a poor/nonexistent jitter
buffer.
I agree with that. Asterisk should just forward any RTP immediately and
let endpoints handle the jitter buffer - unless asterisk is the endpoint
itself (e.g. with phones plugged in its fxs ports).
On Mon, May 01, 2006 at 12:16:18PM +0400, Jean-Michel Hiver spake thusly:
This is only an issue if your SIP phone has a poor/nonexistent jitter
buffer.
I agree with that. Asterisk should just forward any RTP immediately and
let endpoints handle the jitter buffer - unless asterisk is the
Original Message
Skype uses iLBC codec, which has great jitter compensation. IIRC, the
newer SIP channels of * are supposed to have the same capabilities, but
I have not tested. I really do not like Skype (prefer FWD), but I must
say, over satellite, etc, they provide
[EMAIL PROTECTED] wrote:
Original Message
Skype uses iLBC codec, which has great jitter compensation. IIRC, the
newer SIP channels of * are supposed to have the same capabilities, but
I have not tested. I really do not like Skype (prefer FWD), but I must
say, over satellite,
and gave up
eventually.
Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
f(713) 864-8668
c(713) 201-1262
Original Message
Subject: Re: [Asterisk-Users] Compare to Skype
From: Ronald Wiplinger [EMAIL PROTECTED]
Date: Sun, April
[EMAIL PROTECTED] wrote:
What would be ideal is the introduction of an open source wideband codec
implementation. Then you could see it adopted into SIP end points and
used with SER realtively quickly. Sadly, an Asterisk implmentation
would lag a little behind due to the amount of work required
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the
time. Of course his voice quality is like a morse code with dashes or
dots of connection time.
The next minute he calls me via Skype and
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to
transport audio for SIP (and other protocols). This means that ANY
jitter on the SIP Phone - Asterisk link will cause audio problems.
2) Asterisk times it's outgoing audio based on the incoming
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
connection time.
The next minute he calls me via Skype and it
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
connection time.
The next minute he calls me via Skype and it
On Sun, 2006-04-30 at 11:53 +0800, Ronald Wiplinger wrote:
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
13 matches
Mail list logo
