[Asterisk-Dev] Re: Asterisk-Dev digest, Vol 1 #570 - 8 msgs

Mouhamed Mahi SY msy at chaka.sn
Wed Apr 14 05:41:17 MST 2004 

Does anyone try to use TTS  and Voice recognition with Asterisk?
Help me if so


----- Original Message -----
From: <asterisk-dev-request at lists.digium.com>
To: <asterisk-dev at lists.digium.com>
Sent: Wednesday, April 14, 2004 1:42 PM
Subject: Asterisk-Dev digest, Vol 1 #570 - 8 msgs


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> Today's Topics:
>
>    1. Re: Re: 802.11b a contraindication? (Olle E. Johansson)
>    2. Re: Re: 802.11b a contraindication? (Duane)
>    3. Re: Asterisk Codecs Extension (Alex Volkov)
>    4. RE: Asterisk Codecs Extension (Pedro Bessa Goncalves)
>    5. Re: How many users can one Asterisk server support? (Nicolas
Bougues)
>    6. Who is developing Shadydial? (Richard Airlie)
>    7. H263 SIP Video Playback (Pedro Bessa Goncalves)
>    8. Re: Re: 802.11b a contraindication? (Rich Adamson)
>
> --__--__--
>
> Message: 1
> Date: Wed, 14 Apr 2004 08:21:30 +0200
> From: "Olle E. Johansson" <oej at edvina.net>
> Organization: Edvina AB
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] Re: 802.11b a contraindication?
> Reply-To: asterisk-dev at lists.digium.com
>
> > I agree that a more specific understanding of interactions between
> > 802.11[b,a,g] and SIP RTP sessions would be worthwhile if it could be
> > found or generated and posted to this list.  Additionally, what would be
> > more worthwhile would be a similar IAX2 study.   I'll put this on my
> > long and un-cheery list of "things to be tested", right beside satellite
> > latency effects on IAX2...
>
> Some AP's claim they prioritize voice - I don't know how. Symbol have said
> that for a long time, and they supported H.323 in their equipment. There's
> a new QoS standard for 802.11, but I don't know how that works with
various
> protocols or equipments. Check 802.11e.
>
> For more infromation on VOW (voice over WiFi):
>
> http://www.wi-fiplanet.com/tutorials/article.php/2171721
> http://www.vonmag.com/issue/2003/sepoct/features/voice_over_wifi.htm
>
> /O
>
> --__--__--
>
> Message: 2
> Date: Wed, 14 Apr 2004 16:56:03 +1000
> From: Duane <digium at aus-biz.com>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] Re: 802.11b a contraindication?
> Reply-To: asterisk-dev at lists.digium.com
>
> Olle E. Johansson wrote:
> > Some AP's claim they prioritize voice - I don't know how. Symbol have
said
> > that for a long time, and they supported H.323 in their equipment.
There's
> > a new QoS standard for 802.11, but I don't know how that works with
various
> > protocols or equipments. Check 802.11e.
>
> The way I think Cisco was planning to priorities traffic was using VLANs
> that cheat basically...
>
> With the 802.11b timing there is a small slice (1 or 7uS i think, too
> long ago) that isn't currently being used for anything, was set aside
> for QoS or something in the original spec but never used later on, so
> what they do is add this slice to voice traffic's timing to give it a
> higher chance of beating other clients from transmitting...
>
> --
> Best regards,
>   Duane
>
> http://www.cacert.org - Free Security Certificates
> http://www.nodedb.com - Think globally, network locally
> http://www.sydneywireless.com - Telecommunications Freedom
> http://happysnapper.com.au - Sell your photos over the net!
> http://e164.org - Using Enum.164 to interconnect asterisk servers
>
> --__--__--
>
> Message: 3
> From: "Alex Volkov" <avolkov at bpvn.com>
> To: <asterisk-dev at lists.digium.com>
> Subject: Re: [Asterisk-Dev] Asterisk Codecs Extension
> Date: Wed, 14 Apr 2004 03:20:11 -0400
> Reply-To: asterisk-dev at lists.digium.com
>
> This is a multi-part message in MIME format.
>
> ------=_NextPart_000_03F7_01C421CF.65194BC0
> Content-Type: text/plain;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> Asterisk Codecs ExtensionPedro,
>
> Simply look at any codec code in asterisk/codecs/. I suggest =
> codec_ulaw.c or codec_alaw.c. The bulk of both of these is the necessary =
> interfacing code to make whatever-generic-codec work with asterisk, and =
> the actual conversions from xlaw to PCM and back are simple 2-liner =
> loops. Once you understand the basic structure of an * codec, look at =
> codec_g723_1.c for an example on how to handle a whatever-generic-codec =
> that is an opaque library.
> Model your new * codec on those, as I think those provide the best =
> examples of an adaptation/abstraction layer.
>
> Cheers!
> Alex.
>   ----- Original Message -----=20
>   From: Pedro Bessa Goncalves=20
>   To: Asterisk-Dev at Lists. Digium. Com (asterisk-dev at lists.digium.com)=20
>   Sent: Tuesday, April 13, 2004 7:35 AM
>   Subject: [Asterisk-Dev] Asterisk Codecs Extension
>
>
>   Hi, does anyone know the required function headers for extending =
> Asterisk codecs?=20
>   Is there another way to import 3rd party codecs?=20
>
>   Thank you,=20
>   Pedro Goncalves=20
>
> ------=_NextPart_000_03F7_01C421CF.65194BC0
> Content-Type: text/html;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
> <HTML><HEAD><TITLE>Asterisk Codecs Extension</TITLE>
> <META http-equiv=3DContent-Type content=3D"text/html; =
> charset=3Diso-8859-1">
> <META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR>
> <STYLE></STYLE>
> </HEAD>
> <BODY bgColor=3D#ffffff>
> <DIV><FONT face=3DArial size=3D2>Pedro,</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
> <DIV><FONT face=3DArial size=3D2>Simply look at any codec code in =
> asterisk/codecs/.=20
> I suggest codec_ulaw.c or codec_alaw.c. The bulk of both of these is the =
>
> necessary interfacing code to make whatever-generic-codec work with =
> asterisk,=20
> and the actual conversions from xlaw to PCM and back are simple 2-liner =
> loops.=20
> Once you understand the basic structure of an * codec, look at =
> codec_g723_1.c=20
> for an example on how to handle a whatever-generic-codec that is =
> an&nbsp;opaque=20
> library.</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>Model your&nbsp;new * codec on those, =
> as I think=20
> those provide the best examples of an adaptation/abstraction =
> layer.</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
> <DIV><FONT face=3DArial size=3D2>Cheers!</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>Alex.</FONT></DIV>
> <BLOCKQUOTE dir=3Dltr=20
> style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
> BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
>   <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
>   <DIV=20
>   style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
> black"><B>From:</B>=20
>   <A title=3Dest-p-bgoncalves at ptinovacao.pt=20
>   href=3D"mailto:est-p-bgoncalves at ptinovacao.pt">Pedro Bessa =
> Goncalves</A> </DIV>
>   <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
>   title=3Dasterisk-dev at lists.digium.com=20
>   href=3D"mailto:Asterisk-Dev at Lists. Digium. Com =
> (asterisk-dev at lists.digium.com)">Asterisk-Dev at Lists.=20
>   Digium. Com (asterisk-dev at lists.digium.com)</A> </DIV>
>   <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, April 13, 2004 =
> 7:35=20
>   AM</DIV>
>   <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Asterisk-Dev] =
> Asterisk Codecs=20
>   Extension</DIV>
>   <DIV><BR></DIV>
>   <P><FONT size=3D2>Hi, does anyone know the required function headers =
> for=20
>   extending Asterisk codecs?</FONT> <BR><FONT size=3D2>Is there another =
> way to=20
>   import 3rd party codecs?</FONT> </P>
>   <P><FONT size=3D2>Thank you,</FONT> <BR><FONT size=3D2>Pedro =
> Goncalves</FONT>=20
> </P></BLOCKQUOTE></BODY></HTML>
>
> ------=_NextPart_000_03F7_01C421CF.65194BC0--
>
>
> --__--__--
>
> Message: 4
> From: Pedro Bessa Goncalves <est-p-bgoncalves at ptinovacao.pt>
> To: asterisk-dev at lists.digium.com
> Subject: RE: [Asterisk-Dev] Asterisk Codecs Extension
> Date: Wed, 14 Apr 2004 10:57:44 +0100
> Reply-To: asterisk-dev at lists.digium.com
>
> This message is in MIME format. Since your mail reader does not understand
> this format, some or all of this message may not be legible.
>
> ------_=_NextPart_001_01C42206.EF170D30
> Content-Type: text/plain
>
> Hi Alex, what if the codec I want to add is for video? I noticed H263 is
> only available in asterisk/formats/. Does that mean H263 is only being
coded
> in client side and being passed-through by *?
>
> In that case, if I wanted to support the pass-through of a video stream
the
> only thing I would have to do would be changing the SIP header to indicate
> the new codec name?
>
>
>
> Thank you,
>
> Pedro
>
>
>
>   _____
>
> From: Alex Volkov [mailto:avolkov at bpvn.com]
> Sent: quarta-feira, 14 de Abril de 2004 8:20
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] Asterisk Codecs Extension
>
>
>
> Pedro,
>
>
>
> Simply look at any codec code in asterisk/codecs/. I suggest codec_ulaw.c
or
> codec_alaw.c. The bulk of both of these is the necessary interfacing code
to
> make whatever-generic-codec work with asterisk, and the actual conversions
> from xlaw to PCM and back are simple 2-liner loops. Once you understand
the
> basic structure of an * codec, look at codec_g723_1.c for an example on
how
> to handle a whatever-generic-codec that is an opaque library.
>
> Model your new * codec on those, as I think those provide the best
examples
> of an adaptation/abstraction layer.
>
>
>
> Cheers!
>
> Alex.
>
> ----- Original Message -----
>
> From: Pedro Bessa Goncalves <mailto:est-p-bgoncalves at ptinovacao.pt>
>
> To: Asterisk-Dev at Lists. Digium. Com
>
<mailto:Asterisk-Dev at Lists.%20Digium.%20Com%20(asterisk-dev at lists.digium.com
> )>  (asterisk-dev at lists.digium.com)
>
> Sent: Tuesday, April 13, 2004 7:35 AM
>
> Subject: [Asterisk-Dev] Asterisk Codecs Extension
>
>
>
> Hi, does anyone know the required function headers for extending Asterisk
> codecs?
> Is there another way to import 3rd party codecs?
>
> Thank you,
> Pedro Goncalves
>
>
> ------_=_NextPart_001_01C42206.EF170D30
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>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> style=3D'font-size:
> 10.0pt;font-family:Arial;color:navy'>Hi Alex, what if the codec I want =
> to add is
> for video? I noticed H263 is only available in asterisk/formats/. Does =
> that
> mean H263 is only being coded in client side and being passed-through =
> by *?<o:p></o:p></span></font></p>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> style=3D'font-size:
> 10.0pt;font-family:Arial;color:navy'>In that case, if I wanted to =
> support the
> pass-through of a video stream the only thing I would have to do would =
> be
> changing the SIP header to indicate the new codec =
> name?<o:p></o:p></span></font></p>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> style=3D'font-size:
> 10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>=
>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> style=3D'font-size:
> 10.0pt;font-family:Arial;color:navy'>Thank =
> you,<o:p></o:p></span></font></p>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> style=3D'font-size:
> 10.0pt;font-family:Arial;color:navy'>Pedro<o:p></o:p></span></font></p>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> style=3D'font-size:
> 10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>=
>
>
> <div>
>
> <div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
> size=3D3
> face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>
>
> <hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>
>
> </span></font></div>
>
> <p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
> style=3D'font-size:10.0pt;
> font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
> size=3D2
> face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Alex =
> Volkov
> [mailto:avolkov at bpvn.com] <br>
> <b><span style=3D'font-weight:bold'>Sent:</span></b> quarta-feira, 14 =
> de Abril de
> 2004 8:20<br>
> <b><span style=3D'font-weight:bold'>To:</span></b> =
> asterisk-dev at lists.digium.com<br>
> <b><span style=3D'font-weight:bold'>Subject:</span></b> Re: =
> [Asterisk-Dev]
> Asterisk Codecs Extension</span></font><o:p></o:p></p>
>
> </div>
>
> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
> style=3D'font-size:
> 12.0pt'><o:p>&nbsp;</o:p></span></font></p>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial'>Pedro,</span></font><o:p></o:p></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
> style=3D'font-size:
> 12.0pt'>&nbsp;<o:p></o:p></span></font></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial'>Simply look at any codec code in asterisk/codecs/. I =
> suggest
> codec_ulaw.c or codec_alaw.c. The bulk of both of these is the =
> necessary
> interfacing code to make whatever-generic-codec work with asterisk, and =
> the
> actual conversions from xlaw to PCM and back are simple 2-liner loops. =
> Once you
> understand the basic structure of an * codec, look at codec_g723_1.c =
> for an
> example on how to handle a whatever-generic-codec that is =
> an&nbsp;opaque
> library.</span></font><o:p></o:p></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial'>Model your&nbsp;new * codec on those, as I think =
> those
> provide the best examples of an adaptation/abstraction =
> layer.</span></font><o:p></o:p></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
> style=3D'font-size:
> 12.0pt'>&nbsp;<o:p></o:p></span></font></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial'>Cheers!</span></font><o:p></o:p></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial'>Alex.</span></font><o:p></o:p></p>
>
> </div>
>
> <blockquote style=3D'border:none;border-left:solid black =
> 1.5pt;padding:0cm 0cm 0cm 4.0pt;
> margin-left:3.75pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt=
> '>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial'>----- Original Message ----- =
> <o:p></o:p></span></font></p>
>
> </div>
>
> <div style=3D'font-color:black'>
>
> <p class=3DMsoNormal style=3D'background:#E4E4E4'><b><font size=3D2 =
> face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;font-weight:bold'>From:</spa=
> n></font></b><font
> size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;font-family:Arial'> <a
> href=3D"mailto:est-p-bgoncalves at ptinovacao.pt"
> title=3D"est-p-bgoncalves at ptinovacao.pt">Pedro Bessa Goncalves</a> =
> <o:p></o:p></span></font></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><b><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial;font-weight:bold'>To:</span></font></b><font size=3D2
> face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> <a
> href=3D"mailto:Asterisk-Dev at Lists.%20Digium.%20Com%20(asterisk-dev at lists=
> .digium.com)"
> title=3D"asterisk-dev at lists.digium.com">Asterisk-Dev at Lists. Digium. Com
> (asterisk-dev at lists.digium.com)</a> <o:p></o:p></span></font></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><b><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial;font-weight:bold'>Sent:</span></font></b><font =
> size=3D2
> face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> =
> Tuesday, April 13,
> 2004 7:35 AM<o:p></o:p></span></font></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><b><font size=3D2 face=3DArial><span =
> style=3D'font-size:10.0pt;
> font-family:Arial;font-weight:bold'>Subject:</span></font></b><font =
> size=3D2
> face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> =
> [Asterisk-Dev]
> Asterisk Codecs Extension<o:p></o:p></span></font></p>
>
> </div>
>
> <div>
>
> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
> style=3D'font-size:
> 12.0pt'><o:p>&nbsp;</o:p></span></font></p>
>
> </div>
>
> <p><font size=3D2 face=3D"Times New Roman"><span =
> style=3D'font-size:10.0pt'>Hi, does
> anyone know the required function headers for extending Asterisk =
> codecs?</span></font>
> <br>
> <font size=3D2><span style=3D'font-size:10.0pt'>Is there another way to =
> import 3rd
> party codecs?</span></font> <o:p></o:p></p>
>
> <p><font size=3D2 face=3D"Times New Roman"><span =
> style=3D'font-size:10.0pt'>Thank
> you,</span></font> <br>
> <font size=3D2><span style=3D'font-size:10.0pt'>Pedro =
> Goncalves</span></font> <o:p></o:p></p>
>
> </blockquote>
>
> </div>
>
> </body>
>
> </html>
>
> ------_=_NextPart_001_01C42206.EF170D30--
>
> --__--__--
>
> Message: 5
> Date: Wed, 14 Apr 2004 12:02:54 +0200
> From: Nicolas Bougues <nbougues-listes at axialys.net>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] How many users can one Asterisk server
support?
> Organization: Axialys Interactive http://www.axialys.net
> Reply-To: asterisk-dev at lists.digium.com
>
> On Sun, Apr 11, 2004 at 12:57:30AM -0500, Steven Sokol wrote:
> > > The question is
> > > how do we link up these servers and allow any user come in on any
servers?
> > >
> > > Have anyone tried this?
> > >
> >
> > Unfortunately, Asterisk stores the registration information locally,
rather
> > than in some centrally located data store.  In order to use it the way
you
> > want, you either need to alter it to use a central registry, or assign
> > customers to specific servers.
> >
>
> This is not always easily doable, due to NAT issues.
>
> If some SIP phone is behind a NAT gateway, another Asterisk server
> (different from the one having received the registration) won't
> necessarily be able to reach the phone behind the NAT. Thus the whole
> SIP registry shall not be directly shared, but calls should be
> forwarded from one server to another, using existing registration
> info.
>
> Or another, possibly simpler solution is to set users to different
> servers based on their login or something.
>
> --
> Nicolas Bougues
> Axialys Interactive
>
> --__--__--
>
> Message: 6
> Date: Wed, 14 Apr 2004 11:14:38 +0100
> From: Richard Airlie <richard at darq.net>
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] Who is developing Shadydial?
> Reply-To: asterisk-dev at lists.digium.com
>
> I've been looking at the information on Shadydial at
> shadydial.sourceforge.net, but can't find a contact address or even a name
> of the author(s).
>
> I'm interested in finding out a bit more information about how well
shadydial
> performs, what sort of real world usage it's seen, etc. Can anyone shed
any
> light?
>
> best,
> Richard
>
> --__--__--
>
> Message: 7
> From: Pedro Bessa Goncalves <est-p-bgoncalves at ptinovacao.pt>
> To: "Asterisk-Users at Lists. Digium. Com (asterisk-users at lists.digium.com)"
<asterisk-users at lists.digium.com>,
> "Asterisk-Dev at Lists. Digium. Com (asterisk-dev at lists.digium.com)"
<asterisk-dev at lists.digium.com>
> Date: Wed, 14 Apr 2004 11:40:05 +0100
> Subject: [Asterisk-Dev] H263 SIP Video Playback
> Reply-To: asterisk-dev at lists.digium.com
>
> This message is in MIME format. Since your mail reader does not understand
> this format, some or all of this message may not be legible.
>
> ------_=_NextPart_001_01C4220C.D99EC730
> Content-Type: text/plain
>
> Hi. Was anyone able to send an H263 to SIP clients through any Asterisk
play
> function?
> If so, which h263 test files did you use?
>
> Thank you,
> Pedro Goncalves
>
> ------_=_NextPart_001_01C4220C.D99EC730
> Content-Type: text/html
>
> <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
> <HTML>
> <HEAD>
> <META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
> <META NAME="Generator" CONTENT="MS Exchange Server version 5.5.2654.45">
> <TITLE>H263 SIP Video Playback</TITLE>
> </HEAD>
> <BODY>
>
> <P><FONT SIZE=2>Hi. Was anyone able to send an H263 to SIP clients through
any Asterisk play function?</FONT>
> <BR><FONT SIZE=2>If so, which h263 test files did you use?</FONT>
> </P>
>
> <P><FONT SIZE=2>Thank you,</FONT>
> <BR><FONT SIZE=2>Pedro Goncalves</FONT>
> </P>
>
> </BODY>
> </HTML>
> ------_=_NextPart_001_01C4220C.D99EC730--
>
> --__--__--
>
> Message: 8
> Date: Wed, 14 Apr 2004 06:21:52 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: Re: [Asterisk-Dev] Re: 802.11b a contraindication?
> To: asterisk-dev at lists.digium.com
> Reply-To: asterisk-dev at lists.digium.com
>
> > > I agree that a more specific understanding of interactions between
> > > 802.11[b,a,g] and SIP RTP sessions would be worthwhile if it could be
> > > found or generated and posted to this list.  Additionally, what would
be
> > > more worthwhile would be a similar IAX2 study.   I'll put this on my
> > > long and un-cheery list of "things to be tested", right beside
satellite
> > > latency effects on IAX2...
> >
> > Some AP's claim they prioritize voice - I don't know how. Symbol have
said
> > that for a long time, and they supported H.323 in their equipment.
There's
> > a new QoS standard for 802.11, but I don't know how that works with
various
> > protocols or equipments. Check 802.11e.
>
> There are only two common ways that network hardware supports QoS. One
> method has been to watch for the specific IP ports used by sip or h323,
> looking inside those packets to watch for rtp port negotiation, and then
> prioritize the traffic seen on those ports. The cisco pix firewall does
> something like that with their "fixup" statements for allowing access (not
> QoS).
>
> The second common way is to watch the TOS (Type of Service) bits in the IP
> header, and prioritize the traffic based on specific bit patterns. That's
> one typical way to handle QoS in cisco routers as an example.
>
> I'm with Olle in that I don't know what Symbol or others have actually
> implemented, however out of the box there aren't very many network devices
> that truly support QoS in any form. And, in most cases if they do support
> some form of QoS prioritization, their support is not well documented in
> their marketing/sales material or spec sheets.
>
> QoS really does not do much good unless the majority of network devices
> between the voip endpoints all support QoS. In the case of AP's, there is
> already a problem with queuing prior to the voip traffic "reaching" the
> AP from the wireless client (eg, queuing to grab a piece of the wireless
> bandwidth does not involve QoS).
>
> Even if all network devices support QoS, managing the queues is still a
> major operational problem. E.g., what happens when the high priority
> queue is full and additional traffic arrives? How do you know when the
> queue is full and how do you know when to add more bandwidth? I've been
> doing professional network performance analysis for corporations in 40+
> states since 1993, and I've not seen any network support organization
> truly manage their bandwidth or network quality yet, let alone QoS.
>
> If one thinks about how current wireless endpoints control the quality
> of wireless transmission by varying bandwidth, and combine that with how
> one manages the QoS queues, don't think there will be any real
> implementations that actually work using current technology. The current
> stuff does work fine with voip for low usage wireless links, but as that
> traffic increases (or one/two wireless devices hog bandwidth) the voip
> traffic will be impacted one way or another.
>
> Those of us that have analyzed iax and iax2 queuing already know that
> well designed jitter buffers (etc) can handle 500 millisecond latency
> with hugh jitter variations and maintain quality audio. In the short
> term, there is more to be gained in optimizing the jitter buffers then
> there is in truly attempting to manage QoS on an end-to-end network
> basis. (Obviously there are some examples of where QoS can have a
> significant impact though, but most are temporary point solutions.)
>
> Rich
>
>
>
>
> --__--__--
>
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