/mcpRemoteUDPPort=18280 The remote User Datagram Protocol (UDP) port for the call. RoundTripDelay=53 ms The round trip delay as measured by the gateway. SelectedQoS=best-effort The Resource Reservation Protocol (RSVP) is not selected in the dial peer for this call. tx_DtmfRelay=cisco-rtp The form of DTMF RELAY used for the call (if any). SessionProtocol=cisco The Session Protocol for the call. Protocol "cisco" is the default, using H.323 signaling and RTP packets for the voice traffic. Session Intitiation Protocol (SIP) is the other VoIP signaling protocol that can be specified with the help of the session protocol (registered customers only) dial peer command. Non-VoIP protocols such as AAL2 for VoATM or the Cisco proprietary Voice over Frame Relay (VoFR) protocol and FRFll for VoFR can also be specified. SessionTarget=ipv4:10.1.1.2 The session-target from the dial peer. The session target is RAS if a gatekeeper is used. OnTimeRvPlayout=742740 The duration in ms of the voice playout from the data received on time for this call. The Total Voice Playout Duration can be derived by adding the gap fill durations to the OnTimeRvPlayout duration. GapFillWithSilence=0 ms Time (ms) Gateway (GW) played silence. Silence plays out in these situations: • When a packet is lost and there is no audio sample available to play. For example, when two or more packets are lost in sequence. This situation can result in an audible click or gap being heard by the user. • When the playout buffer adapts to a larger value by inserting silence between buffered voice packets. This situation does not result in an audible loss in quality. GapFillWithPrediction=0 ms The duration in ms of the voice signal played out with the signal synthesized from parameters, or samples of data that precede it in time. This gap fill occurs because voice data is lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser and frame-concealment strategies in G.729 and G.723.1 compression algorithms. GapFillWithInterpolation=0 ms As for GapFillWithPrediction but taking into consideration samples received after the missing voice traffic and stored in the de-jitter buffer. Not currently used. GapFillWithRedundancy=0 ms If a redundant encoding scheme is used by the transmitter, then the payload of lost or late packets can be partially or fully recovered and played out with a reduced impact on voice quality. This technique is not currently supported. HiWaterPlayoutDelay=70 ms The First-In, First-Out (FIFO) jitter buffer high mark that indicates the maximum depth to which the de-jitter buffer adapts for this call.