Studio Interoperability

Radio Studio

Summary: Streaming audio over IP (AoIP) networks is being increasingly used by broadcasting companies, among others, to provide high-quality audio feeds over distance across an IP network such as the Internet.

When it comes to Studio-to-Studio links, the interoperability is of particular importance. It can be a very complex  task to achieve interoperability, because of the wide variety of Audio over IP products that connect with the studio. There are microphones, mixers, codecs, sound processors and much more devices that interact through IP, but 2wcom is committed to maintain interoperability to various devices by supporting different standards and adapting to the fast changes of the IP world.

The EBU TECH 3326 was one of the first standards that ensured a common transmission method between Audio over IP devices. From this standard evolved several other standards like AES67, RAVENNA, Livewire+ and Dante.

All of these implementations are very common when it comes to the used technologies.

1. Session Initiation (SIP)

The session initiation can be compared to calling a destination via telephone (in fact SIP is widely used for IP telephony).  The client has to register itself with a  “Registrar” on the SIP server. Once registered it can be called by other codecs using the SIP uri. 2wcom has integrated this protocol so neatly that the user can setup entire phone books inside of the device (MM01) and call other codecs with a single click:

2. Session Description (SDP)

The session description is used to negotiate a certain format that both codecs will agree on and use during their session. It is defined in the Session Description Protocol (SDP), which is used by the session initiation (SIP) or session announcement (SAP). It is very important to implement the mandatory SDP profiles defined in the AES67 standard to be interoperable with other codec manufacturers. 

3. Codecs

The more codecs a device supports the more compatible it will be with other devices. Here is a list of codecs that are supported by most 2wcom devices:

  • PCM, 16-24 Bit, 32kHz – 192kHz
  • MPEG 1/2 Layer 1, 2, 3
  • AAC (LC, LD, HEv1, HEv2)
  • enhanced aptX
  • G.711
  • G.722
  • [optional] Opus
  • [optional] AC3
4. Realtime Streaming

Our implementation uses the well known protocols RTP and RTSP to provide a standardized realtime streaming for audio. Beside these protocols techniques like Dual Streaming and Pro-MPEG FEC are used to improve the reliability to a rock solid operation. Customer needs, for example GPIO forwarding, expands the feature set of the 2wcom devices to a state-of-the-art level.

While all of the above helps with network insufficiencies, RTSP (Real-Time Streaming Protocol) makes the streaming as easy as using a media player. With the functions DESCRIBE, SETUP, RECORD, PAUSE, PLAY, TEARDOWN and more our audio over ip codecs can be controlled as if they were a media player.