Wireshark-bugs: [Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP sig

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From: bugzilla-daemon@xxxxxxxxxxxxx
Date: Wed, 30 Nov 2016 20:33:50 +0000

Comment # 1 on bug 13193 from 
Created attachment 15096 [details]
TLS 1.2 and DTLS 1.0 master secrets (SSLKEYLOGFILE) for dtls-srtp-ws-sip.pcapng

For this test I used "DTMF demo" (from https://sip5060.net/test-calls/):
https://freephonebox.net/?dial=test.dtmf%40sip5060.net&video=false
which uses Asterisk 11.13.1.

Only the TCP stream for WSS/SIP and UDP (STUN, DTLS-SRTP/G.711, DNS) is
preserved to reduce capture size.

For Opus, there is also https://webrtc.freeswitch.org/

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