Which protocol should be analyzed when troubleshooting call quality between IP phones?
When addressing call quality issues between IP phones, the Real-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.
In contrast:
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.