From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/pc/g3doc/srtp.md | 72 ++++++++++++++++++++++++++++++++++ 1 file changed, 72 insertions(+) create mode 100644 third_party/libwebrtc/pc/g3doc/srtp.md (limited to 'third_party/libwebrtc/pc/g3doc/srtp.md') diff --git a/third_party/libwebrtc/pc/g3doc/srtp.md b/third_party/libwebrtc/pc/g3doc/srtp.md new file mode 100644 index 0000000000..eb457efacf --- /dev/null +++ b/third_party/libwebrtc/pc/g3doc/srtp.md @@ -0,0 +1,72 @@ + + + +# SRTP in WebRTC + +WebRTC mandates encryption of media by means of the Secure Realtime Protocol, or +SRTP, which is described in +[RFC 3711](https://datatracker.ietf.org/doc/html/rfc3711). + +The key negotiation in WebRTC happens using DTLS-SRTP which is described in +[RFC 5764](https://datatracker.ietf.org/doc/html/rfc5764). The older +[SDES protocol](https://datatracker.ietf.org/doc/html/rfc4568) is implemented +but not enabled by default. + +Unencrypted RTP can be enabled for debugging purposes by setting the +PeerConnections [`disable_encryption`][1] option to true. + +## Supported cipher suites + +The implementation supports the following cipher suites: + +* SRTP_AES128_CM_HMAC_SHA1_80 +* SRTP_AEAD_AES_128_GCM +* SRTP_AEAD_AES_256_GCM + +The SRTP_AES128_CM_HMAC_SHA1_32 cipher suite is accepted for audio-only +connections if offered by the other side. It is not actively supported, see +[SelectCrypto][2] for details. + +The cipher suite ordering allows a non-WebRTC peer to prefer GCM cipher suites, +however they are not selected as default by two instances of the WebRTC library. + +## cricket::SrtpSession + +The [`cricket::SrtpSession`][3] is providing encryption and decryption of SRTP +packets using [`libsrtp`](https://github.com/cisco/libsrtp). Keys will be +provided by `SrtpTransport` or `DtlsSrtpTransport` in the [`SetSend`][4] and +[`SetRecv`][5] methods. + +Encryption and decryption happens in-place in the [`ProtectRtp`][6], +[`ProtectRtcp`][7], [`UnprotectRtp`][8] and [`UnprotectRtcp`][9] methods. The +`SrtpSession` class also takes care of initializing and deinitializing `libsrtp` +by keeping track of how many instances are being used. + +## webrtc::SrtpTransport and webrtc::DtlsSrtpTransport + +The [`webrtc::SrtpTransport`][10] class is controlling the `SrtpSession` +instances for RTP and RTCP. When +[rtcp-mux](https://datatracker.ietf.org/doc/html/rfc5761) is used, the +`SrtpSession` for RTCP is not needed. + +[`webrtc:DtlsSrtpTransport`][11] is a subclass of the `SrtpTransport` that +extracts the keying material when the DTLS handshake is done and configures it +in its base class. It will also become writable only once the DTLS handshake is +done. + +## cricket::SrtpFilter + +The [`cricket::SrtpFilter`][12] class is used to negotiate SDES. + +[1]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/peer_connection_interface.h;l=1413;drc=f467b445631189557d44de86a77ca6a0c3e2108d +[2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/media_session.cc;l=297;drc=3ac73bd0aa5322abee98f1ff8705af64a184bf61 +[3]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=33;drc=be66d95ab7f9428028806bbf66cb83800bda9241 +[4]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=40;drc=be66d95ab7f9428028806bbf66cb83800bda9241 +[5]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=51;drc=be66d95ab7f9428028806bbf66cb83800bda9241 +[6]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=62;drc=be66d95ab7f9428028806bbf66cb83800bda9241 +[7]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=69;drc=be66d95ab7f9428028806bbf66cb83800bda9241 +[8]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=72;drc=be66d95ab7f9428028806bbf66cb83800bda9241 +[9]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=73;drc=be66d95ab7f9428028806bbf66cb83800bda9241 +[10]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_transport.h;l=37;drc=a4d873786f10eedd72de25ad0d94ad7c53c1f68a +[11]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/dtls_srtp_transport.h;l=31;drc=2f8e0536eb97ce2131e7a74e3ca06077aa0b64b3 +[12]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_filter.h;drc=d15a575ec3528c252419149d35977e55269d8a41 -- cgit v1.2.3