fec79b74a8
BUG=None No-Try: true Change-Id: I2677c1e932e2a4e0833f7c3185689ab030c8fa61 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218608 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34127}
49 lines
4.0 KiB
Markdown
49 lines
4.0 KiB
Markdown
<?% config.freshness.reviewed = '2021-05-13' %?>
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<?% config.freshness.owner = 'hta' %?>
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# SRTP in WebRTC
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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).
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The key negotiation in WebRTC happens using DTLS-SRTP which is described in [RFC 5764](https://datatracker.ietf.org/doc/html/rfc5764).
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The older [SDES protocol](https://datatracker.ietf.org/doc/html/rfc4568) is implemented but not enabled by default.
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Unencrypted RTP can be enabled for debugging purposes by setting the PeerConnections [`disable_encryption`][1] option to true.
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## Supported cipher suites
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The implementation supports the following cipher suites:
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- SRTP_AES128_CM_HMAC_SHA1_80
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- SRTP_AEAD_AES_128_GCM
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- SRTP_AEAD_AES_256_GCM
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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.
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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.
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## cricket::SrtpSession
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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.
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Encryption and decryption happens in-place in the [`ProtectRtp`][6], [`ProtectRtcp`][7], [`UnprotectRtp`][8] and [`UnprotectRtcp`][9] methods.
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The `SrtpSession` class also takes care of initializing and deinitializing `libsrtp` by keeping track of how many instances are being used.
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## webrtc::SrtpTransport and webrtc::DtlsSrtpTransport
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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.
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[`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.
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## cricket::SrtpFilter
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The [`cricket::SrtpFilter`][12] class is used to negotiate SDES.
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[1]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/peer_connection_interface.h;l=1413;drc=f467b445631189557d44de86a77ca6a0c3e2108d
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[2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/media_session.cc;l=297;drc=3ac73bd0aa5322abee98f1ff8705af64a184bf61
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[3]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=33;drc=be66d95ab7f9428028806bbf66cb83800bda9241
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[4]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=40;drc=be66d95ab7f9428028806bbf66cb83800bda9241
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[5]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=51;drc=be66d95ab7f9428028806bbf66cb83800bda9241
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[6]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=62;drc=be66d95ab7f9428028806bbf66cb83800bda9241
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[7]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=69;drc=be66d95ab7f9428028806bbf66cb83800bda9241
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[8]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=72;drc=be66d95ab7f9428028806bbf66cb83800bda9241
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[9]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_session.h;l=73;drc=be66d95ab7f9428028806bbf66cb83800bda9241
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[10]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_transport.h;l=37;drc=a4d873786f10eedd72de25ad0d94ad7c53c1f68a
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[11]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/dtls_srtp_transport.h;l=31;drc=2f8e0536eb97ce2131e7a74e3ca06077aa0b64b3
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[12]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/srtp_filter.h;drc=d15a575ec3528c252419149d35977e55269d8a41
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