When addressing call quality issues between IP phones, theReal-time Transport Protocol (RTP)is the primary protocol to analyze. RTP is responsible for the actual transmission of voice data during a call, and its performance directly impacts call quality.
Packet Loss:Missing packets can lead to audio gaps.
Jitter:Variations in packet arrival times can cause choppy audio.
Latency:Delays in packet delivery can result in noticeable lag.
Analyzing RTP streams allows technicians to identify these issues and implement appropriate Quality of Service (QoS) measures to mitigate them.
Incontrast:
SIP (Session Initiation Protocol):Handles call setup, modification, and teardown but not the media stream.
SCCP (Skinny Client Control Protocol):A Cisco proprietary protocol for signaling, not media transport.
H.323:An older protocolsuite for multimedia communication, encompassing both signaling and media, but less commonly used in modern IP telephony.Reference:Supporting Cisco Devices for Field Technicians (FLDTEC) – Troubleshooting Methodologie
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