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Codec Exchange During the Setup Phase of an Internet Protocol (IP) Phone Call

IP.com Disclosure Number: IPCOM000123688D
Original Publication Date: 1999-Mar-01
Included in the Prior Art Database: 2005-Apr-05
Document File: 2 page(s) / 96K

Publishing Venue

IBM

Related People

Truong, HL: AUTHOR [+5]

Abstract

In the following, methods will be described, which permit the introduction and use of new codecs without requiring an upgrade of the signalling software stack, i.e. without the need of standardization.

This text was extracted from an ASCII text file.
This is the abbreviated version, containing approximately 52% of the total text.

Codec Exchange During the Setup Phase of an Internet Protocol (IP)
Phone Call

   In the following, methods will be described, which permit
the introduction and use of new codecs without requiring an upgrade
of the signalling software stack, i.e. without the need of
standardization.

   One advantage that IP telephony has over the conventional
one is its flexibility regarding the selection of the codec to be
used during a phone call.  While in the conventional cases (PSTN,
ISDN, the same codec (namely G.711) is  used for all calls, a
multitude of codecs is defined in IP telephony.  H.323 terminals for
example exchange during the call setup phase the identity of the
encoders and decoders (H.245 capabilities set exchange) they
support.  Knowing the receive-capabilities of a peer, a terminal can
then select the appropriate encoder for the media streams it will
send to that peer.  With such a negotiation mechanism not only
different codecs can be selected for different calls, but also
different codecs can be selected for the forward and backward
directions.

   A similar codec negotiation mechanism is also defined in
Session Initiation Protocol (SIP).

   Despite of the flexibility described above, the codec
selection can only succeed in cases both terminals support a common
codec.  That is why in H.323 a codec, namely G.711, is prescribed as
mandatory to ensure interoperability.

   Another problem arises when a new codec with better
characteristics (e.g. better voice quality, better compression
ratio, etc.) is introduced.  The new codec can only be used if all
terminals involved in a call
  1.  have installed the codec, and
  2.  have upgraded their signalling protocol stacks
      for the support of the new codec.

   Condition 1 can be relatively easy implemented, in that
the terminals download the new codec from somewhere in the Internet
and install it locally.  Condition 2 is more difficult to fulfill,
since it not only requires the installation of a signalling software
upgrade but also the standardization of the necessary protocol
elements to enable the identification of the new codec.  The
standardization process is however time-consuming and therefore slows
down the introduction pace of new codecs.

   The subsequently described methods permit the introduction
and use of new codecs without the need of an upgrade of the
signalling software stack.

  ...