RTP Packet in VoIP Call

asked 2020-04-17 09:22:59 +0000

Hilman gravatar image

updated 2020-04-17 09:55:21 +0000

grahamb gravatar image

Hello, I am trying to test the encryption process in VoIP communication. Before, I try capture "audio in VoIP" communication "without encryption".

When I try to capture VoIP audio between "two clients in one bridge", the RTP packet only appears as many as "4 pieces" in the main display of Wireshark. When "one client stops communication" and the other client is still on the bridge, the RTP packet then appears until the client stops communication.

Are there certain settings that must be met so that Wireshark can capture VoIP audio between two clients who are communicating? Anybody can help ?

Note, Wireshark is run on the same computer as the communication server. Thanks

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Have you looked at the signaling too? It could be that after the first few RTP packets, the audio between the two VoIP endpoints is sent directly (which means your capture point is not in the path of the RTP packets)

SYN-bit gravatar imageSYN-bit ( 2020-04-17 09:53:34 +0000 )edit

I think signaling is appropriate. I also tried to filter packets that were captured based on the ip device used in communication, but still did not display data with the RTP packet while communicating (2 channel in 1 bridge). When one channel remains in the bridge, the RTP packet then appears until the channel is hanged.

Hilman gravatar imageHilman ( 2020-04-17 10:36:57 +0000 )edit