[asterisk-users] TCP port, VPN and resolving the cutting voice problem

[Mike]

I'm certainly not a network expert (beyond normal network knowledge for an
IT person), and I agree with you TCP SHOULD be dropped first because of what
you said, but I often heard so-called network expert (or at least some
holding jobs as such) say that it`s normal my UDP packets get dropped
because they are UDP.

 

UDP is, or was until recently, considering to be carrying those "who cares?"
packets, like SNMP, etc. Sort of like ICMP.  The reverse logic held true,
where if it was a mechanism using UDP, it was because nobody really cared
about reliability in the first place.

 

VoIP may have changed some people`s mind, of course. 

 

Mike

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Jones
Sent: Tuesday, November 30, 2010 1:33 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] TCP port, VPN and resolving the cutting voice
problem

 

Just the contrary - Most QoS schemes will drop TCP first, specifically
because it is known that with TCP, the packet will be resent, so no
application will be hurt.  UDP is not dropped first because it is known that
there will likely be more impact.

I am not aware of any way to run IAX over TCP, and I agree it would be a bad
idea.  The proper thing to do is to implement PROPER QoS on BOTH SIDES of
the link, which I hope is point to point.  If it goes over the Internet,
your QoS is lost as soon as it hits a router you dont control (or pay for
QoS services on)

I think in IAX, the jitter buffer size can be adjusted, but I dont know the
detail on this..  A jitter buffer can be looked upon as like a funnel - as
packets arrive, they are dumped in the funnel, which is then pouring your
audio out the bottom in a smooth stream, no matter how much delay there is
in the filling of the funnel.   When the funnel runs out of packets (ie:
delay has caused you to run out of data) then you get a break in your audio
stream.  Increasing the jitter buffer (bigger funnel) can fix this, but at a
certain point, the audio will be SO DELAYED (because the packets are waiting
in the buffer) that the users will notice and get confused.



-Steve



---------original message ------

From: "Mike" <list at net-wall.com>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Date: Tue, 30 Nov 2010 12:34:08 -0500
Subject: Re: [asterisk-users] TCP port, VPN and resolving the cutting voice
problem

> I know understand the latency due to the resending .. But if the link was
have a good speed internet, then resending will make a big latency?

I think the point is that with TCP, congestion will create a vicious circle
of more congestion, while with UDP congestion is bad in itself, but doesn't
result in more congestion created by the original congestion.

That being said, isn't UDP sometimes looked at as being lower priority than
TCP by many routers out there and dropped first when congestion does occur?
That makes it a good reason to use TCP in some cases.

Mike

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