[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

Ivan Nardi  changed:

   What|Removed |Added

 CC||nardi.i...@gmail.com

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[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

Peter Wu  changed:

   What|Removed |Added

 Status|IN_PROGRESS |CONFIRMED

--- Comment #6 from Peter Wu  ---
This is now listed at
https://wiki.wireshark.org/GSoC2020#WebRTC:_DTLS-SRTP_decryption_and_Opus_playback_support

Further notes:
The original WebRTC architecture link is unfortunately dead, however the
details are still the same.

DTLS-SRTP uses the exporter secret as defined in
Keying Material Exporters for Transport Layer Security (TLS)
https://tools.ietf.org/html/rfc5705#section-4

The Opus codec library can be found at:
https://opus-codec.org/

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[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

--- Comment #5 from Gerrit Code Review  ---
Change 19139 had a related patch set uploaded by Peter Wu:
sdp: decode pt for more RTP transport protocols

https://code.wireshark.org/review/19139

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[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

--- Comment #4 from Peter Wu  ---
The SDP dissector could use some refactoring to make extensions easier
(https://code.wireshark.org/review/19047). References for further refinement:

SDP: Session Description Protocol
https://tools.ietf.org/html/rfc4566

An Offer/Answer Model with the Session Description Protocol (SDP)
https://tools.ietf.org/html/rfc3264

Session Initiation Protocol (SIP) Usage of the Offer/Answer Model
https://tools.ietf.org/html/rfc6337

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[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

Jaap Keuter  changed:

   What|Removed |Added

 Status|UNCONFIRMED |IN_PROGRESS
 Ever confirmed|0   |1

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[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

--- Comment #3 from Gerrit Code Review  ---
Change 18997 had a related patch set uploaded by Peter Wu:
SDP: add basic ICE candidate attribute dissection (RFC 5245)

https://code.wireshark.org/review/18997

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[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

--- Comment #2 from Gerrit Code Review  ---
Change 18996 had a related patch set uploaded by Peter Wu:
DTLS: add support for use_srtp extension (RFC 5764)

https://code.wireshark.org/review/18996

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[Wireshark-bugs] [Bug 13193] Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets

[bugzilla-daemon]

https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=13193

--- Comment #1 from Peter Wu  ---
Created attachment 15096
  --> https://bugs.wireshark.org/bugzilla/attachment.cgi?id=15096=edit
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=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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