[sip-comm] Voice quality


#1

This is a general question for all users ... does anyone know how many users or d/l's of SIP communicator there are, and what the general consensus is on the quality of the voicepath? We have tried many times using ADSL on one end and cable modem on the other, with Windows XP machines with decent processors and memory, and we have not been able to get a satisfactory quality two-way audio stream established. We can establish a reasonable quality in one direction, but the other direction always gets clipped and drops packets, is delayed, and is otherwise unintelligible.

We are wondering if others have experienced this. It seems more like a network bandwidth/QoS or JMF problem vs a SIP Communicator problem, but we just wanted to find out other people's experiences.

Martin

···

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#2

Hello,

I also tried the Sip communicator under windows XP and it seems that there is always a delay from the receiver to the caller.
The shame is that it was on a local network which was very little (2 PCs and a switch), so I guess the loss of quality is not coming from the distance not long Internet routes.
I did not used any option nor customisation and I will be quite happy to know that I did not use a correct option enabling me to have a better upward quality.

Regards,
St�phane

Martin Barclay wrote:

···

This is a general question for all users ... does anyone know how many users or d/l's of SIP communicator there are, and what the general consensus is on the quality of the voicepath? We have tried many times using ADSL on one end and cable modem on the other, with Windows XP machines with decent processors and memory, and we have not been able to get a satisfactory quality two-way audio stream established. We can establish a reasonable quality in one direction, but the other direction always gets clipped and drops packets, is delayed, and is otherwise unintelligible.

We are wondering if others have experienced this. It seems more like a network bandwidth/QoS or JMF problem vs a SIP Communicator problem, but we just wanted to find out other people's experiences.

Martin

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#3

Hi all:

Yes, we've also experienced this. This is a JMF problem and nothing much can
be done about it. Actually, JMF is not usable as a VoIP endpoint media
solution. The delay is more than a second and quite unacceptable.

I think the community should think about replacing JMF with some other media
engine altogether. The other option is to use as much native as possible
with JMF (since JMF is actually a framework and custom high-performance
implementations are possible). The default implementation of JMF is just not
usable for VoIP.

We can think of starting an open-source initiative, may be OpenJMF. We can
wrap existing C/C++ based media engine components available in public domain
and open source into JMF.

Without enhancements on the media performance side, SIP-Communicator will
always remain an experimental interest only and will never be used actually.

Regards,
Shafqat

···

-----Original Message-----
From: Martin Barclay [mailto:martin@martinbarclay.com]
Sent: Friday, February 25, 2005 10:13 AM
To: users@sip-communicator.dev.java.net
Subject: [sip-comm] Voice quality

This is a general question for all users ... does anyone know how many
users or d/l's of SIP communicator there are, and what the general
consensus is on the quality of the voicepath? We have tried many times
using ADSL on one end and cable modem on the other, with Windows XP
machines with decent processors and memory, and we have not been able
to get a satisfactory quality two-way audio stream established. We can
establish a reasonable quality in one direction, but the other
direction always gets clipped and drops packets, is delayed, and is
otherwise unintelligible.

We are wondering if others have experienced this. It seems more like a
network bandwidth/QoS or JMF problem vs a SIP Communicator problem, but
we just wanted to find out other people's experiences.

Martin

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#4

I agree ... Yet JMF is far from being the essential reason for the sip-communicator being in an experimental phase. The project is still quite young and (though real close to a point of maturity) there are quite some glitches like for example reliability and deployment that need to be addressed first. We count to do that during the next 6 months. A few persons are about to join the project and that's what we are goin to concentrate on first.

As for JMF - yup I agree that it's not perfect. Yet if used properly it could produce decent results and serve as a temprary solution. The current implementation could not possibly allow SIP Communicator and other JMF VoIP applications to become the best in the field ... yet it is good enough to earn their keep until sth better is available.

Which means that we (the open source community) have the time to work on sth better. How realistic is that? I don't know. Would it actually be done? It better be ...

Emil

Shafqat Ullah wrote:

···

Hi all:

Yes, we've also experienced this. This is a JMF problem and nothing much can
be done about it. Actually, JMF is not usable as a VoIP endpoint media
solution. The delay is more than a second and quite unacceptable.

I think the community should think about replacing JMF with some other media
engine altogether. The other option is to use as much native as possible
with JMF (since JMF is actually a framework and custom high-performance
implementations are possible). The default implementation of JMF is just not
usable for VoIP.

We can think of starting an open-source initiative, may be OpenJMF. We can
wrap existing C/C++ based media engine components available in public domain
and open source into JMF.

Without enhancements on the media performance side, SIP-Communicator will
always remain an experimental interest only and will never be used actually.

Regards,
Shafqat

-----Original Message-----
From: Martin Barclay [mailto:martin@martinbarclay.com]
Sent: Friday, February 25, 2005 10:13 AM
To: users@sip-communicator.dev.java.net
Subject: [sip-comm] Voice quality

This is a general question for all users ... does anyone know how many
users or d/l's of SIP communicator there are, and what the general
consensus is on the quality of the voicepath? We have tried many times
using ADSL on one end and cable modem on the other, with Windows XP
machines with decent processors and memory, and we have not been able
to get a satisfactory quality two-way audio stream established. We can
establish a reasonable quality in one direction, but the other
direction always gets clipped and drops packets, is delayed, and is
otherwise unintelligible.

We are wondering if others have experienced this. It seems more like a
network bandwidth/QoS or JMF problem vs a SIP Communicator problem, but
we just wanted to find out other people's experiences.

Martin

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#5

Hi,

I have found a BUG (???) in JMF.

If you don't make:

//
player.getVisualComponent();
player.getControlPanelComponent();
//

you don't have any DELAY ( < 1 second);

try, and you will see different ...

Wbr,

···

On Fri, Feb 25, 2005 at 04:36:13PM +0100, Emil Ivov wrote:

I agree ... Yet JMF is far from being the essential reason for the
sip-communicator being in an experimental phase. The project is still
quite young and (though real close to a point of maturity) there are

Emil

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