Which protocol should be analyzed when troubleshooting call quality between IP phones?

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.​

Key factors affecting RTP and, consequently, call quality include:​

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