![]() Wireshark Tips and Tricks for RTP Analysis ![]() A MOS score above 3.5 is usually considered satisfactory for voice calls. Another factor is the MOS (Mean Opinion Score), a subjective measure of voice quality ranging from 1 (poor) to 5 (excellent). A one-way latency below 150 ms is generally considered acceptable for VoIP applications. High latency, or the time it takes for a packet to travel from sender to receiver, can also affect call quality. Additional Indicators of Poor RTP ConnectionsĪside from packet loss and jitter, there are other indicators of poor RTP connections. A jitter value below 30 ms is typically considered acceptable for voice traffic. This will display a detailed graph of jitter values over time. In Wireshark, you can analyze jitter by selecting an RTP stream from the Telephony > RTP > RTP Streams window and clicking the "Analyze" button. Jitter refers to the variation in packet arrival times, which can cause audio distortion and reduced call quality. A packet loss rate below 1% is generally considered acceptable for voice traffic. This will display detailed information about all RTP streams, including packet loss percentages. To identify packet loss in Wireshark, use the Telephony > RTP > RTP Streams menu option. It occurs when packets are dropped or lost during transmission, leading to audio artifacts and degradation in call quality. Packet loss is one of the most common issues affecting RTP voice streams. This will help you focus only on the RTP traffic and exclude other irrelevant packets. To do this, you can use Wireshark's capture filter feature by entering udp portrange 16384-32767 (commonly used RTP port range) in the capture filter field. Capturing RTP Voice Streams in Wiresharkīefore we can analyze RTP voice streams, we need to capture them. In this article, we will focus on how you can analyze RTP voice streams using Wireshark, identify common problems like packet loss and jitter, and understand the acceptable values for these parameters. ![]() Real-time Transport Protocol (RTP) is commonly used for transmitting voice and video data over IP networks, making it a crucial component of many VoIP and video conferencing applications. ![]()
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