SIP for compatibility of IP Audio Codecs
The European Telecommunications Standards Institute (ETSI) was founded 1998, and already one year after there was the DSS1 standard (so called Euro-ISDN) introduced. However it was just the start of the long long process. Audio transmissions over ISDN with professional quality using devices of different manufacturers from different countries was only possible after the long and intensive negotiations between the manufacturers and various standardizing boards, as well as after the number of standards’ and implementations’ improvements.
To avoid the same situation during the establishment of the IP-based connections as the significant communication type for broadcasting industry’s purposes, there was the N/ACIP working group of the EBU founded. The group is already issued the draft recommendations on the audio connections over IP. Based on those drafts MAYAH started the implementations of the newest Audio-over-IP Solutions in its products.
One of the main technology to avoid the mentioned incompatibility difficulties should be the Session Initiation Protocol (SIP). This protocol was standardized by Internet Engineering Taskforce (IETF) and it generally allows the problem-free connections over IP-routes using devices of different manufacturers. Solutions based on SIP are already proved its availability in the Voice-over-IP and telephony industry.
Since there are different algorithms being used in the broadcasting and in the telephony applications, it is important to ensure that the algorithms and their parameters are applied in a standardized form. The N/ACIP working group has already created the general regulations which should be published Autumn 2007.
Manufacturers, who take those regulations in consideration by their developments, allow the smooth communications of their devices to the products of the other companies or to the IP-phones. The codecs of the Audio-via-IP Experts Group members (see also www.audio-via-ip.com) can already communicate to each other using their SIP functionalities.
An establishing of an IP-based audio connection using SIP usually looks like that:
It is quite common to provide more than one alternative encoding so that Codec 2 has several choices. The list of alternatives can be found inside the SDP part of a SIP message and is ordered by priority.
The following shows an excerpt of an example SIP message:
m=audio 5004 RTP/AVP 96 97 9 8 0
a=fmtp:97 layer=2; samplerate=48000; bitrate=384000; mode=Stereo