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Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED

"Fabio Pietrosanti (naif)" <lists@infosecurity.ch> Wed, 27 June 2012 18:22 UTC

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Date: Wed, 27 Jun 2012 20:22:46 +0200
From: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>
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To: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 6/27/12 8:14 PM, Roy, Radhika R CIV (US) wrote:
> Hi, all:
> 
> Real-time protocols (Audio/Video payloads for two-way conversations) over UDP/IP were being developed based on the single thing: Retransmission delays are unacceptable (more over congestion problems especially for the high-bandwidth videos).
> 
> If this fundamental performance characteristic for the Real-time protocols (Audio/Video payloads for two-way conversations) need to be changed, the experimental results with lots of actual measurements must be produced to the technical communities.
> 
> Otherwise, someone may make some subjective statements here and there without any actual measurements will only provide evidences what those folks do not know what they are actually talking about creating technical confusions.

It's not a subjective topic that over a mobile network there is packet
loss and that there are known condition where this lead to unacceptable
voip quality.

The Internet Measurement of VoIP on different Transport Layer Protocols
http://netarchlab.tsinghua.edu.cn/PersonalMainPage/wzl/public_html/pubs/2009_ICOIN_transport_measure.pdf

SipDroid: An opensource Mobile client using RTP Retransmission to
improve quality over 3G network with packet loss:
http://code.google.com/p/sipdroid/wiki/NewImprovedAudio

RTP Retransmission RFC (explaining how to handle dynamically
retransmission from RTCP feedback):
http://tools.ietf.org/html/rfc4588

Fabio

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