如何基于GStreamer重流已保存的RTP原始文件以模拟摄像头RTP流？

Got it, since you already extracted the raw G.729 data from Wireshark and use GStreamer in your project, here's how to replay that stream over UDP for your testing needs—we'll cover two common scenarios based on what exactly you exported from Wireshark:

Scenario 1: You exported pure G.729 audio frames (no RTP headers) #

This is the typical case when you selected the "Raw" option for the audio stream export. Use this GStreamer pipeline to package the frames into RTP packets and send them over UDP:

gst-launch-1.0 filesrc location=/path/to/your/g729_raw_exported_file do-timestamp=true ! audio/x-G729, rate=8000, channels=1 ! rtpg729pay pt=18 ! udpsink host=YOUR_TEST_HOST_IP port=YOUR_TEST_UDP_PORT sync=true

Let's break down what each part does:

• filesrc: Reads your local raw G.729 file
• do-timestamp=true: Ensures the source adds accurate timing metadata to each frame, so we can replay at real-time speed
• audio/x-G729: Explicitly sets the media format—G.729 uses 8kHz sampling rate and mono audio by default, match this to your original captured stream if it differs
• rtpg729pay: Packages the G.729 frames into RTP packets; pt=18 sets the payload type to match what you filtered in Wireshark (rtp.p_type==18)
• udpsink: Sends the RTP packets to your test system's IP and UDP port; sync=true ensures playback matches the original stream's timing

Scenario 2: You exported full RTP packets (including RTP headers) #

If you accidentally exported the complete RTP packets (instead of just the payload), you can skip the RTP packaging step and send the raw packets directly:

gst-launch-1.0 filesrc location=/path/to/your/full_rtp_packets_file do-timestamp=true ! application/x-rtp, payload=18, media=audio, clock-rate=8000 ! udpsink host=YOUR_TEST_HOST_IP port=YOUR_TEST_UDP_PORT sync=true

Additional Tips for Realistic Testing #

• RTCP Support: For more accurate RTP session simulation (including feedback), use rtpbin to handle both RTP and RTCP:

  gst-launch-1.0 rtpbin name=rtpbin \
  filesrc location=/path/to/your/g729_raw_file do-timestamp=true ! audio/x-G729, rate=8000, channels=1 ! rtpg729pay pt=18 ! rtpbin.send_rtp_sink_0 \
  rtpbin.send_rtp_src_0 ! udpsink host=YOUR_TEST_HOST_IP port=YOUR_RTP_PORT \
  rtpbin.send_rtcp_src_0 ! udpsink host=YOUR_TEST_HOST_IP port=YOUR_RTCP_PORT sync=false async=false \
  udpsrc port=YOUR_LOCAL_RTCP_PORT ! rtpbin.recv_rtcp_sink_0
  

• Verify the Stream: On your test system, you can use GStreamer to listen and decode the stream to confirm it works:

  gst-launch-1.0 udpsrc port=YOUR_TEST_UDP_PORT ! application/x-rtp, payload=18, media=audio, clock-rate=8000 ! rtpg729depay ! avdec_g729 ! autoaudiosink
  

内容的提问来源于stack exchange，提问作者krezus