Subject: [dcp] Congestion Control for VoIP

From: Stefan Håkansson (EPL)<[email-masked]>
Date: Fri, 22 Feb 2002 09:23:07 +0100

Hi,

I would like to start with saying that I am sorry for my ignorance, since I am sure this topic has already been extensively treated. However, I would like to know, if there anywhere is an ongoing discussion or conclusion on congestion control for VoIP within the IETF.

VoIP usually involves some sort of speech codec that produces a new frame once every 10, 20 or 30 ms. And in order to have a good conversational quality the round-trip time cannot be too long, this often forces the application to pack only one speech frame in every packet. To slow-start does not really make sense when the service requires a new packet to be sent every 20 (or 10 or 30) ms , at least not when the round-trip time is long. And to halve the bit rate in case of a packet loss is not really a possibility if a low-bit rate speech codec is used.

So in summary, I would be very grateful for any advice on where congestion control for VoIP has been discussed within the IETF, and for sources of information on this topic. 

Best Regards,
Stefan Håkansson                 Ericsson AB
[email-masked]

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---

From: "Krithivas, Ramamurthy" <[email-masked]>
Date: Fri, 22 Feb 2002 08:59:51 -0800

Stefan,

  A starting point would be IETF's Audio-Video Transport Working Group.
Suggest that 
  you take a look at the RTP (Real Time Protocol, RFC1889) draft. 

  I "believe" that the function of RTCP (Real Time Control Protocol) is to
provide 
  feedback (congestion control) functionality for the RTP (streams).  

  http://www.ietf.org/rfc/rfc1889.txt
  http://www.ietf.org/rfc/rfc1890.txt

R> Krithivas

-----Original Message-----
From: Stefan Håkansson (EPL) [mailto:[email-masked]]
Sent: Friday, February 22, 2002 2:23 AM
To: [email-masked]'
Subject: [dcp] Congestion Control for VoIP

Hi,

I would like to start with saying that I am sorry for my ignorance, since I
am sure this topic has already been extensively treated. However, I would
like to know, if there anywhere is an ongoing discussion or conclusion on
congestion control for VoIP within the IETF.

VoIP usually involves some sort of speech codec that produces a new frame
once every 10, 20 or 30 ms. And in order to have a good conversational
quality the round-trip time cannot be too long, this often forces the
application to pack only one speech frame in every packet. To slow-start
does not really make sense when the service requires a new packet to be sent
every 20 (or 10 or 30) ms , at least not when the round-trip time is long.
And to halve the bit rate in case of a packet loss is not really a
possibility if a low-bit rate speech codec is used.

So in summary, I would be very grateful for any advice on where congestion
control for VoIP has been discussed within the IETF, and for sources of
information on this topic. 

Best Regards,