2019年5月2日 · A common failure code is the “RTP/RTCP Timeout,” which indicates a handset on an active call is no longer able to detect media (RTP) for more than 10 to 30 seconds. This occurs when one person can no longer hear the other, but the parties don’t hang up, hoping to reconnect.

2019年12月24日 · 当空口（无线接口）发生丢包时，VoLTE的RTP（Real-time Transport Protocol）包会丢失，导致语音质量下降。由于每个字大约需要8到10个RTP包，持续丢包1秒可能会使用户错过5个字，严重影响通话感知。典型的例子显示...

2023年3月23日 · 通过wireshark打开后看到都是udp流量，需要对数据包进行解码，根据源端口或目的端口 进行decode as（解码）成应用层协议RTP，另外下面举了数据包丢包和数据包乱序的例子。

2014年7月28日 · This document describes the process of how to decipher the Real-Time Streaming (RTP) stream for packet loss analysis in Wireshark for voice and video calls. You can use Wireshark filters in order to analyze simultaneous packet captures taken at or close-to the source and destination of a call.

2020年12月8日 · 通过测试复现及抓Log分析，此问题是由于网络在20秒内没有下行RTP数据包，导致监控RTP数据包的定时器超时，被叫主动中断VoLTE通话链路。 经过分析，该问题是网络原因导致，非终端问题。

2018年6月26日 · Once you have the capture available filter the captures for RTP packets. Then go to Telephony>RTP >RTP Stream. This will pop up following window which will tell you the percentage of RTP packet drops between the nodes.

We have a Cisco 3750 running C3750 Software (C3750-IPBASE-M), Version 12.2 (25)SEE2, RELEASE SOFTWARE (fc1). We have a lot of voice traffice passing through this switch and through sniffing we can see that it is dropping a lots of RTP packets. It is pretty basic configured - no QoS etc defined, no DSCP, WRED etc.

2020年2月27日 · When traffic is finally captured and opened via Wireshark, proceed with troubleshooting the RTP streams. As next steps, select Telephony -> RTP -> RTP Streams. As observed, there are 4 RTP streams, but the first and third one have almost 4% packet loss. Select one of them and then select 'Analyze'.

2024年10月22日 · These are VoLTE drops during SRVCC (if the core does not support these procedures). Let me explain, how this works in a simple manner. If my 4G to 3G IRAT Handover threshold is -118 dBm RSRP and my SRVCC threshold is -110 dBm RSRP then consider a UE at the RSRP of -112 dBm will be on LTE using data only.

2015年3月12日 · A source drops offline and no longer sends any RTCP reports. The source will be timed out and removed from the participant database; A single RTCP report is dropped. This is not enough to timeout the source and RTCP processing will resume on the next received report.