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Cisco Meeting Server Release 2.0 : API Reference Guide
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<txVideoMute>false</txVideoMute>
<sipMediaEncryption>allowed</sipMediaEncryption>
<audioPacketSizeMs>XXXXXX</audioPacketSizeMs>
<deactivationMode>disconnect</deactivationMode>
<deactivationModeTime>1</deactivationModeTime>
<telepresenceCallsAllowed>false</telepresenceCallsAllowed>
<sipPresentationChannelEnabled>true</sipPresentationChannelEnable
d>
<bfcpMode>1</bfcpMode>
</configuration>
A.1.2 Call leg status
The status information returned contains live values relating to the call leg’s active state. Media
information is contained within one or more rxAudio, txAudio, rxVideo and txVideo sub-
sections; multiple video or audio streams may be identified and distinguished via their “role”
attribute, which may be either “main” or “presentation”.
Each media section includes packet loss percentage and jitter. An example set of status data is
shown below:
<status>
<state>connected</state>
<durationSeconds>349</durationSeconds>
<direction>incoming</direction> 
<sipCallId>6ff025ec40843df0b5153cab4a8601ee</sipCallId>
<groupId>8b29e92d-27c2-421a-8d73-47daf09d7fe7</groupId>
<recording>true</recording>
<deactivated>false</deactivated>
<encryptedMedia>true</encryptedMedia>
<unencryptedMedia>false</unencryptedMedia>
<layout>telepresence</layout>
<activeLayout></activeLayout>
<availableVideoStreams></availableVideoStreams>
<rxAudio>
<codec>aac</codec>
<packetLossPercentage>0.0</packetLossPercentage>
<jitter>2</jitter>
<bitRate>64000</bitRate>
</rxAudio>
<txAudio>
<codec>aac</codec>
<packetLossPercentage>0.0</packetLossPercentage>
<jitter>0</jitter>
<bitRate>64000</bitRate>
<roundTripTime>66</roundTripTime>
</txAudio>
<rxVideo role="main">
<codec>h264</codec>
<width>768</width>
Appendix A   Additional Call Leg Information