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When was RTP Buffering - Frame Based Buffering introduced?


In Song module version 8 a new RTP buffering method called frame based buffering was introduced. The algorithm calculates the audio buffer level in milliseconds rather than in bytes.


Frame based buffering allows:

  • configurable decoding delay with one frame accuracy
  • synchronisation of several decoders to the same stream (just by configuring them to the same initial delay)
  • stable delay over long period of time
  • automatic correction of clock difference between encoder and decoder


The following applications use frame based buffering:

Application NameVersion
Streaming Client2.17
Annuncicom Full Duplex0.21


The only configuration parameter for the RTP decoder is the delay in milliseconds.

The delay parameter is the desired processing delay of the decoder (between the network input and the audio output). Please note that the end-to-end delay between the encoder and the decoder might be (significantly) different to the value configured.

In an ideal case the delay parameter would be 0 ms, however due to device's internal buffers a small delay (depending on the hardware) is inevitable. The delay value should also cover possible temporary network hick-ups (jitter). E.g. if the network sometimes delays the packet delivery by 20ms due to a temporary load, the configured parameter should not be less than 20ms.

The maximum configurable delay is limited by the device's internal buffer (64, 32 or 16kB).

Recommended Settings

The following table lists recommended delay values for various audio formats. The value includes 2-frame jitter and is independent on hardware/software.

Audio formatDelay
uLaw/ALaw 8kHz mono444ms
PCM 8kHz mono444ms
uLaw/ALaw 12kHz mono316ms
PCM 12kHz mono316ms
uLaw/ALaw 24kHz mono188ms
PCM 24kHz mono188ms
uLaw/ALaw 32kHz mono156ms
PCM 32kHz mono152ms
PCM 44.1kHz mono110ms
PCM 44.1kHz stereo79ms
PCM 48kHz stereo72ms

Maximum Settings

This section explains the minimum and the maximum delay values for different audio formats and platforms.

The hardware is divided into two groups:

  • Micronas (MAS) based devices: Annuncicom 100/155/200/1000, Exstreamer 1000
  • VLSI based devices: Exstreamer 100/110/200


The following table shows the minimum and the maximum possible delay with MP3 constant bitrate. The maximum delay differs between the Streaming Client, which has 64kB audio buffer available, and ABCL (Annuncicom FDX, STL), which features only 32kB buffer. The minimum delay includes 100ms network jitter.

MP3 CBR bitrateMin delayMax delay (SC)Max delay (ABCL)


Variable or average bitrate the minimum and delay depends on the bitrate variation interval. The minimum delay is taken from the CBR table for the low end of the interval, whereas the maximum delay is the CBR value for the high end of the interval.

Please note that most MP3 encoders use the whole bitrate range starting from the lowest bitrate 32kbps. E.g. VBR 128kbps varies from 32 to 128kbps

MP3 FormatMin delayMax delay (SC)Max delay (ABCL)


In uncompressed audio (PCM, uLaw or ALaw) the minimum and maximum delay depend on the bit rate and on the hardware.

The following table lists minimum and maximum settings for all standard RTP audio formats:

FormatMin delay MASMin delay VLSIMax delay (SC)Max delay (ABCL)Max delay (ABCL full duplex)
uLaw 8kHz mono

ALaw 8kHz mono

PCM 8kHz mono60ms424ms4075ms2027ms1003ms
uLaw 12kHz mono

ALaw 12kHz mono

PCM 12kHz mono54ms296ms2710ms1345ms662ms
uLaw 24kHz mono

ALaw 24kHz mono

PCM 24kHz mono47ms168ms1345ms662ms321ms
uLaw 32kHz mono

ALaw 32kHz mono

PCM 32kHz mono43ms134ms1005ms493ms237ms
PCM 44.1kHz mono31ms97ms729ms358ms172ms
PCM 44.1kHz stereo16ms72ms364ms179ms86ms
PCM 48kHz stereo15ms66ms335ms164ms79ms

Multiple Device Synchronisation

Multiple devices receiving the same RTP stream can be configured to play in sync by entering the same delay parameter.

Barix recommends to use broadcast or multicast together with synchronisation, otherwise a small inaccuracy (few milliseconds) might be caused by the network delivery to different locations.

Deliberate Delays

In some applications it is desired to artificially delay the audio. E.g. in a tunnel to eliminate the delay caused by the distance between the devices.

An artificial delay can be introduced by configuring the devices to different delay values. E.g. 100ms, 120ms, 140ms, 160ms, etc.

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