73 lines
4.1 KiB
Markdown
73 lines
4.1 KiB
Markdown
|
<?% config.freshness.reviewed = '2021-05-13' %?>
|
||
|
|
<?% config.freshness.owner = 'hta' %?>
|
||
|
|
|
||
|
|
# 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
|