Avaya 4600 Series Nota de Aplicação Página 53

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The Audio Push Type
Issue 1 April 2005 53
Note:
Note: For example, if an audio stream originator wanted to explicitly stop an audio
stream the following would be sent in new Push Content:
Audio quality depends on the streaming source providing the audio at an appropriate pace. The
pace depends on the packet size. If you are using 40ms packets, then the packets should be
separated by 40ms.
Transmitting the packets too slowly results in odd silences when the telephone has no audio to
play out its Speaker. Transmitting the packets too quickly results in broken sound, as the
telephone is forced to drop packets it cannot maintain in its buffer.
The error in the pace measured at the telephone is called jitter. If the jitter stays below the size
of one packet, then jitter does not impact the audio quality. Note that a +2ms jitter on one packet
cancels out a -2ms jitter on the next. However, 40ms packets, each separated by 38ms, means
that the jitter grows +2ms with each packet. After 3 seconds, the jitter would be +150ms, and
the telephone would need to drop audio to maintain its buffers.
<!– Following is the XML Push Request Message sent as a POST request embedded as part of form data -->
XMLData = <?xml version="1.0"?>
<Push type=“audio” mode=“barge”>
<go href=“http://trusted_push_server/stop_audio.xml” method=“get”>
</go>
</Push>
<!–The message below is from the Push Content file called “stop_audio.xml” from above go href URI -->
<?xml version="1.0"?>
<Response>
<Audio>
<Url href=”RTPRx://STOP”>
</Audio> <
</Response>
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