Subject: reduce jitter in routed network for voip applications

From: "Daniele Giordano" <[email-masked]>
Date: Sun, 27 Mar 2005 19:58:31 +0200

voip applications use connectionless layer 4 protocols and this increases
jitter. A layer 4 connection oriented protocol limits jitter but it must not
use packet checks in voip applications.
Is there an intefrace between TCP and UDP? Isn't there a "TDP" (transfer
datagram protocol) that joins the two features?
What do you think about this?

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From: Dean Anderson <[email-masked]>
Date: Sun, 27 Mar 2005 19:00:14 -0500 (EST)

> use packet checks in voip applications.

Connection oriented protocols don't limit jitter. Jitter is the same no
matter what. However, a very late packet is discarded by a jitter buffer.  
(you don't care what joe said 10 seconds ago--if it didn't get here in the
jitter buffer, its too late.)  But it would not be discarded by a
connection oriented protocol. You'd hear what joe said 10 seconds ago, and
then you would have to wait 10 seconds to hear what joe just said. This is
OK for streaming a song. Its not fine for two way voice.

> What do you think about this?

Right protocol for the right purpose.  We already have RTSP for streaming, 
and RTP for voip.

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From: "Daniele Giordano" <[email-masked]>
Date: Mon, 28 Mar 2005 13:09:34 +0200

RTP is transparent at the transport layer. We analyse TCP and UDP:
TCP is connection oriented and so the communication begins with the
definition of a virtual circuit.
A virtual circuit is a temporary connection of sequence nodes with relative
reservation of bandwidth.
A connection oriented service gives the certainty that all information units
use the same nodes with a same medium latency.
Same latency maintains reduced the jitter.

UDP is connection less protocol and so routing and forwarding functions are
executed for any information unit in arrival to a node.
Information units with the same destination address can use different path
and so the latency can be different.
A variable delay increases the jitter.

I think that RTP should use a layer 4 connection oriented protocol (like
TCP) but without retransmissions of information units with excessive delay
or errors (like UDP).

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From: Brian E Carpenter <[email-masked]>
Date: Mon, 28 Mar 2005 13:33:32 +0200

Daniele,

Perhaps you should study the differentiated services architecture,
RFC 2475. This is what many deployments of VoIP are using.

     Brian

Daniele Giordano wrote:
>>>voip applications use connectionless layer 4 protocols and this
> 
> increases
> 
>>>jitter. A layer 4 connection oriented protocol limits jitter but it must
> 
> not
>

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From: [email-masked] (Mike S)
Date: Mon, 28 Mar 2005 06:40:29 -0500

At 06:09 AM 3/28/2005, Daniele Giordano wrote...
>Same latency maintains reduced the jitter.

That is incorrect unless _other_, stateful protocols (ex. RSVP, integrated services) are in use throughout the entire network. That is not the general case.

IP routes on a per hop basis, whether TCP or UDP. There is no "virtual circuit" created, except at the endpoints.

>What do you think about this?

You're trying to solve a problem which is incorrectly defined, and therefore doesn't exist.

Diffserv already provides a QoS mechanism for VoIP and does not require gateways to maintain state for each connection. It, like Intserv, cannot be assumed to exist through any random Internet path. RFC2474, RFC2475.

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From: "Daniele Giordano" <[email-masked]>
Date: Mon, 28 Mar 2005 20:26:54 +0200

CoS, ToS, QoS work on the information units queued in a buffer of a device.
I think that, in the same way of a circuit switched network, an
establishment phase could exist. In this phase the network could reserve the
right resources.
i.e. virtual circuit, bandwidth, ....

Thanks.

----- Original Message ----- 
From: "Mike S" <[email-masked]>
To: "Daniele Giordano" <[email-masked]>
Cc: <[email-masked]>
Sent: Monday, March 28, 2005 1:40 PM
Subject: Re: reduce jitter in routed network for voip applications

> That is incorrect unless _other_, stateful protocols (ex. RSVP, integrated
services) are in use throughout the entire network. That is not the general
case.
>
> IP routes on a per hop basis, whether TCP or UDP. There is no "virtual
circuit" created, except at the endpoints.
> Diffserv already provides a QoS mechanism for VoIP and does not require
gateways to maintain state for each connection. It, like Intserv, cannot be
assumed to exist through any random Internet path. RFC2474, RFC2475.

---

From: Hans Kruse <[email-masked]>
Date: Mon, 28 Mar 2005 17:47:50 -0500

That establishment/reservation phase is RSVP; as others have said, 
available in theory, but cannot be assumed to exist in the network in 
general.

On Mar 28, 2005, at 13:26, Daniele Giordano wrote:

>>> A connection oriented service gives the certainty that all 
>>> information
> units
Hans Kruse, Associate Professor
J. Warren McClure School of Communication Systems Management
Adjunct Associate Professor of Electrical Engineering and Computer 
Science
292 Lindley Hall, Ohio University, Athens, OH, 45701
740-593-4891 voice, 740-593-4889 fax

---

From: Greg Daley <[email-masked]>
Date: Tue, 29 Mar 2005 11:11:21 +1000

Hi Daniele,

> Same latency maintains reduced the jitter.

[chop udp discussion]
> What do you think about this?

TCP does induce delay variation when retransmission is considered.

End-to-end retransmission of VoIP is not useful in many cases.  TCP
by default does this.

The additional mechanisms in TCP to throttle transmissions when
multiple duplicate ACKs are seen will reduce the rate of transmission
in some cases for VoIP.  This will also cause packet delay variation.

TCP is not well suited to VoIP for these reasons.

Additionally, UDP and TCP packets travel through the network
at the same rate (if no differential forwarding is used).
The premise that TCP and UDP have different properties on the forwarding
plane is thus flawed.

Greg

---

From: Dean Anderson <[email-masked]>
Date: Tue, 29 Mar 2005 11:10:03 -0500 (EST)

> Same latency maintains reduced the jitter.

The variable latency is usually introduced as a result of queuing due to
transient congestion in the path, and to a lesser extent, due to path
variations due to load balancing.  It has (almost) nothing to do with
whether the packets are TCP or UDP.  QOS protocols may alter the queueing
priorities, but QOS should be helping to reduce VOIP latency by pushing
the VOIP packets to the front of the transmission queue.

Extensive testing done at Genuity around 2000 showed that congestion is
mainly a problem in the tail circuit and that low latency queuing (and I
think smaller MTU) allowed good RTP latencies right up to full t1
utilization. Partly, this solution is specific to Genuity, which had (I
think)  larger than typical headroom in the core.

RSVP won't really do much unless you have end-to-end implementation--I
won't go into any criticisms of RSVP, but suffice it to say I think its
still a work in progress (good work, but more needed), and as a practical
matter isn't widely implemented in the general internet and isn't likely
to be widely implemented between carriers, I think. RSVP as it stands is
really only useful to some fortune 500 companies that have their own
large, private, wide area networks.  QOS in the core networks is
impractical for a number of reasons.  Building in excess headroom is a
good idea for a voip network.  A voip-only side-network is also a good
idea.

I don't think TCP is a solution to latency. Indeed, that would possibly
subject voice streams to RED and WRED congestion control.  (these drop tcp
packets randomly to cause windowing reductions to reduce packet rates
which then reduces congestion)

> What do you think about this?

What connection oriented features would you add besides retransmission and 
windowing? You said that your'd remove retransmission, and windowing is 
already essentially implmented in the jitter buffer.

One could say that (voip) RTP is already connection-oriented with H323 and
SIP handling the RTP connection setup issues. Usually, this is necessary
because the h323 and SIP functions are intimately tied to call routing and
stream setup.

		--Dean