![]() 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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