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	<title>Comments on: How To: Basic QoS on Vyatta with DSCP for VoIP</title>
	<atom:link href="http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/</link>
	<description>Just another geek</description>
	<pubDate>Fri, 30 Jul 2010 21:54:18 +0000</pubDate>
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		<title>By: mwatson</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-52</link>
		<dc:creator>mwatson</dc:creator>
		<pubDate>Thu, 28 Aug 2008 13:05:12 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-52</guid>
		<description>Maillikarjun,

Sorry I have yet to attempt setting up BGP on Vyatta, you might want to checkout the forums on http://www.vyatta.org however, you may find some advice there.</description>
		<content:encoded><![CDATA[<p>Maillikarjun,</p>
<p>Sorry I have yet to attempt setting up BGP on Vyatta, you might want to checkout the forums on <a href="http://www.vyatta.org" rel="nofollow">http://www.vyatta.org</a> however, you may find some advice there.</p>
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		<title>By: mallikarjun</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-50</link>
		<dc:creator>mallikarjun</dc:creator>
		<pubDate>Mon, 25 Aug 2008 15:15:50 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-50</guid>
		<description>How to configure the bgp protocol in 3 vyatta router</description>
		<content:encoded><![CDATA[<p>How to configure the bgp protocol in 3 vyatta router</p>
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		<title>By: mwatson</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-44</link>
		<dc:creator>mwatson</dc:creator>
		<pubDate>Wed, 06 Aug 2008 15:46:40 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-44</guid>
		<description>Hi Kevin,

I would suspect your echo problem is not being caused by LAN congestion problem - especially if you are running on gigabit.

I also don;t believe an improper QoS problem would cause echo'ing either... I would suspect that poor QoS policies may cause distortion/static or voices "cutting in and out".

I believe your problem is probably the TalkSwitch system.  I'm not familiar with the TalkSwitch products.  

Does your problem only occur on calls involving your analog lines?  More than likely what you are hearing is what is called "Hybrid Echo", this is echo generated by a device in telco switches called a "Hybrid" basically what the Hybrid does is convert your single-pair copper analog connection into a 2-pair connection.  The Hybrid does this to seperate the send &#038; recieve signals.  The connection between the Telco switch and your location is always delivered on a single pair - the send and receive is mixed together.  The telco seperates the send and recieve at their end so that these signals can be easily digitized, compressed, and trunked over a much larger pipe.  The problem is, if the echo is not properly tuned for your specific line impedence, it creates echo due to electrical leakage.  

When your phone line is connected to a regular analog handset, this echo is not normally a problem because the echo comes back so quickly to the person speaking that it just sounds like "sidetone" to them (they can hear themselves talking in the earpiece).  The problem here is... as soon as you connect that analog line to a VoIP system, the time your VoIP system takes to digitize the analog signal, send it over your IP network to your phone handset, the phone then re-converts it back to analog and out the earpiece.  The amount of time that takes makes you actually hear the echo as an echo of yourself, instead of your brain interpreting it as sidetone.

Like I said, I can't speak for the TalkSwitch products, I am using an Asterisk based system.

Asterisk, and the underlying Analog interface drivers (ZapTel / DAHDI) provide ways for you to adjust the gains on your lines which can drastically reduce the echo caused by the Hybrid.  Essentially you are tuning for the line impedance on your end of the connection instead of relying on the Telco to do it (which most don't because its not a problem to 99% of their customers using pure analog systems).

Some ZapTel / DAHDI analog interface cards also come with hardware-based echo cancellors built directly onto the card to help reduce echo.  In place of a hardware echo cancellor, ZapTel / DAHDI also offer a half dozen or so different software-based echo cancellation algorithms.

I would imagine that TalkSwitch has some type of similiar method to tuning the lines.  Its a little bit of a task to go through and tune all of your lines - especially if you have alot of analog lines since they all need to be done individually.

I know this doesn;t apply to TalkSwitch, but you may find reading this post about how I did the line tuning with Asterisk available here: http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/

Good Luck!</description>
		<content:encoded><![CDATA[<p>Hi Kevin,</p>
<p>I would suspect your echo problem is not being caused by LAN congestion problem - especially if you are running on gigabit.</p>
<p>I also don;t believe an improper QoS problem would cause echo&#8217;ing either&#8230; I would suspect that poor QoS policies may cause distortion/static or voices &#8220;cutting in and out&#8221;.</p>
<p>I believe your problem is probably the TalkSwitch system.  I&#8217;m not familiar with the TalkSwitch products.  </p>
<p>Does your problem only occur on calls involving your analog lines?  More than likely what you are hearing is what is called &#8220;Hybrid Echo&#8221;, this is echo generated by a device in telco switches called a &#8220;Hybrid&#8221; basically what the Hybrid does is convert your single-pair copper analog connection into a 2-pair connection.  The Hybrid does this to seperate the send &#038; recieve signals.  The connection between the Telco switch and your location is always delivered on a single pair - the send and receive is mixed together.  The telco seperates the send and recieve at their end so that these signals can be easily digitized, compressed, and trunked over a much larger pipe.  The problem is, if the echo is not properly tuned for your specific line impedence, it creates echo due to electrical leakage.  </p>
<p>When your phone line is connected to a regular analog handset, this echo is not normally a problem because the echo comes back so quickly to the person speaking that it just sounds like &#8220;sidetone&#8221; to them (they can hear themselves talking in the earpiece).  The problem here is&#8230; as soon as you connect that analog line to a VoIP system, the time your VoIP system takes to digitize the analog signal, send it over your IP network to your phone handset, the phone then re-converts it back to analog and out the earpiece.  The amount of time that takes makes you actually hear the echo as an echo of yourself, instead of your brain interpreting it as sidetone.</p>
<p>Like I said, I can&#8217;t speak for the TalkSwitch products, I am using an Asterisk based system.</p>
<p>Asterisk, and the underlying Analog interface drivers (ZapTel / DAHDI) provide ways for you to adjust the gains on your lines which can drastically reduce the echo caused by the Hybrid.  Essentially you are tuning for the line impedance on your end of the connection instead of relying on the Telco to do it (which most don&#8217;t because its not a problem to 99% of their customers using pure analog systems).</p>
<p>Some ZapTel / DAHDI analog interface cards also come with hardware-based echo cancellors built directly onto the card to help reduce echo.  In place of a hardware echo cancellor, ZapTel / DAHDI also offer a half dozen or so different software-based echo cancellation algorithms.</p>
<p>I would imagine that TalkSwitch has some type of similiar method to tuning the lines.  Its a little bit of a task to go through and tune all of your lines - especially if you have alot of analog lines since they all need to be done individually.</p>
<p>I know this doesn;t apply to TalkSwitch, but you may find reading this post about how I did the line tuning with Asterisk available here: <a href="http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/" rel="nofollow">http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/</a></p>
<p>Good Luck!</p>
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		<title>By: Kevin</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-43</link>
		<dc:creator>Kevin</dc:creator>
		<pubDate>Tue, 05 Aug 2008 18:15:27 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-43</guid>
		<description>We have been trying to diagnose our internal VOIP extension problem: echoing, etc.  Our external lines are analog, and internally we are going through a gigabit switch to the router, and back to our phone system (talkswitch) so I would not believe there would be any saturation.  

I'm not sure what the problem is yet, but I'm wondering if a QOS policy is something that we should be looking to establish on our vyatta router. Thoughts?</description>
		<content:encoded><![CDATA[<p>We have been trying to diagnose our internal VOIP extension problem: echoing, etc.  Our external lines are analog, and internally we are going through a gigabit switch to the router, and back to our phone system (talkswitch) so I would not believe there would be any saturation.  </p>
<p>I&#8217;m not sure what the problem is yet, but I&#8217;m wondering if a QOS policy is something that we should be looking to establish on our vyatta router. Thoughts?</p>
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		<title>By: mwatson</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-29</link>
		<dc:creator>mwatson</dc:creator>
		<pubDate>Mon, 30 Jun 2008 13:34:02 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-29</guid>
		<description>Hi Joel,

I'm glad you are finding the guide useful!

In my setup here the connections to the datacenter are infact leased lines.  However it should work in your situation of using VPN's as well.. though it may need some tweaking depending on your setup.</description>
		<content:encoded><![CDATA[<p>Hi Joel,</p>
<p>I&#8217;m glad you are finding the guide useful!</p>
<p>In my setup here the connections to the datacenter are infact leased lines.  However it should work in your situation of using VPN&#8217;s as well.. though it may need some tweaking depending on your setup.</p>
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		<title>By: Joel</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-28</link>
		<dc:creator>Joel</dc:creator>
		<pubDate>Mon, 30 Jun 2008 01:20:13 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-28</guid>
		<description>This is an excellent article Matt.  Thanks a lot. I am using this as a guide for policies for my VoIP softphone connecting through an SSL VPN client back to the Cisco CM.

Based on the diagram these sites are VPN'd into the datacenter over the internet? or are they leased telco connections?</description>
		<content:encoded><![CDATA[<p>This is an excellent article Matt.  Thanks a lot. I am using this as a guide for policies for my VoIP softphone connecting through an SSL VPN client back to the Cisco CM.</p>
<p>Based on the diagram these sites are VPN&#8217;d into the datacenter over the internet? or are they leased telco connections?</p>
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		<title>By: John Babio</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-27</link>
		<dc:creator>John Babio</dc:creator>
		<pubDate>Fri, 27 Jun 2008 11:39:41 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-27</guid>
		<description>Awesome!! I have been meaning to mess with the vyatta QoS Stuff.</description>
		<content:encoded><![CDATA[<p>Awesome!! I have been meaning to mess with the vyatta QoS Stuff.</p>
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		<title>By: Matt Watson&#8217;s Blog Is Interesting &#171; Clearing My Head</title>
		<link>http://www.mattgwatson.ca/2008/05/how-to-basic-qos-on-vyatta-with-dscp-for-voip/#comment-14</link>
		<dc:creator>Matt Watson&#8217;s Blog Is Interesting &#171; Clearing My Head</dc:creator>
		<pubDate>Mon, 09 Jun 2008 23:25:10 +0000</pubDate>
		<guid isPermaLink="false">http://www.mattgwatson.ca/?p=16#comment-14</guid>
		<description>[...] parts, of migrating his employer to an Asterisk based PBX. He also has some nice description of establishing QoS for VOIP using the Vyatta open source router. I&#8217;m really curious about Vyatta, but I do think its perhaps more than a little beyond my [...]</description>
		<content:encoded><![CDATA[<p>[...] parts, of migrating his employer to an Asterisk based PBX. He also has some nice description of establishing QoS for VOIP using the Vyatta open source router. I&#8217;m really curious about Vyatta, but I do think its perhaps more than a little beyond my [...]</p>
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