Concluded WG Privacy Enhanced RTP Conferencing (perc)
Note: The data for concluded WGs is occasionally incorrect.
|WG||Name||Privacy Enhanced RTP Conferencing|
|Area||Applications and Real-Time Area (art)|
|Additional resources||Issue tracker, Wiki|
|Personnel||Chairs||Nils Ohlmeier, Suhas Nandakumar|
|Area Director||Murray Kucherawy|
Final Charter for Working Group
RTP-based real-time multi-party interactive media conferencing is in widespread use today. Many of the deployments use one or more centrally located media distribution devices that perform selective forwarding of mixed-media streams received from the participating endpoints. The media transport protocol commonly used is RTP (RFC3550). There are various signaling systems used to establish these multi-party conferences.
These conferences require security to ensure that the RTP media and related metadata of the conference is kept private and only available to the set of invited participants and other devices trusted by those participants with their media. At the same time, multi-party media conferences need source authentication and integrity checks to protect against modifications, insertions, and replay attacks. Media distribution devices supporting these conferences may also perform RTP header changes, and they often consume and create RTCP messages for efficient media handling.
To date, deployment models for these multi-party media distribution devices do not enable the devices to perform their functions without having keys to decrypt the participants' media, primarily using Secure RTP (RFC3711) to provide session security.
This trust model has limitations and prevents or hampers deployment of secure RTP conferencing in a multitude of cases, including outsourcing, legal requirements on confidentiality, and utilization of virtualized servers. Thus, a new architectural model and related specifications are needed, with a focused effort from the RTP and Security communities.
This WG will work on a solution that enables centralized SRTP-based conferencing, where the central device distributing the media is not required to be trusted with the keys to decrypt the participants' media. The media must be kept confidential and authenticated between an originating endpoint and the explicitly allowed receiving endpoints or other devices. The meta information provided to the central device is to be limited to the minimal required for it to perform its function to preserve the conference participants privacy. Further, it is desired that a solution still provides replay protection, so that the media distribution devices cannot replay previous parts of the media.
The solution must also provide security for each hop between endpoints and multi-party media distribution devices and between multi-party media distribution devices. The RTCP messages and RTP header extensions required for the media distribution device to perform the selective media forwarding may require both source authentication and integrity as well as confidentiality. The solution may also consider providing end-to-end security for a subset of the RTCP messages or RTP header extensions.
The solution should be implementable by both SIP (RFC3261) and WebRTC endpoints (draft-ietf-rtcweb-overview). How telepresence endpoints using the protocols defined in the CLUE working group could utilize the defined security solution needs to be considered. However, it is acknowledged that limitations may exist, resulting in restricted functionality or need for additional adaptations of the CLUE protocols.
This working group will perform the following work:
- Define a general architecture and RTP topology(s) that enables end-to-end media security for multi-party RTP conferencing.
- Define the trust model and describe the resulting security properties.
- Document models considered for integrating the solution with WebRTC, SIP and CLUE establishment of conferencing sessions.
- Specify any necessary extensions to SRTP.
- Define needed Key Management Functions that distribute the keys. The system needs to be able to bind the media to the identity of the sender of the media and/or the identity of the conference.
If there is identification of missing protocols or functionalities, such work can be requested to be done in another working group with a suitable charter or by requests for chartering it in this WG or another WG. Potential items that might require work in other WGs are DTLS extensions (TLS WG) and RTP header extensions (AVTEXT WG). This work requires strong collaboration with the security area. We will notify, and when needed coordinate with, AVTCore, CLUE, MMUSIC, RTCWEB, SIPREC, W3C WebRTC, and other related groups about this work.
The PERC working group is not chartered to extend any signaling system used to establish the RTP-based conferences. It will, however, need to consider in its architecture how the solution may integrate with these systems, and document such consideration for WebRTC, SIP, and CLUE. The specification of how a particular system integrates the security solution will happen outside of this working group, in suitable venues.
The WG will not consider non-real-time usage, multicast-based media distribution, or Security descriptions-based keying (RFC4568).
Input to the WG
Proposals already existing relating to this charter proposal:
|Mar 2018||Submit documentation of how to integrate solution in SIP, WebRTC and CLUE to IESG (Informational)|
|Dec 2017||Submit Key-management protocol specification to IESG (Standards Track)|
|Dec 2017||Submit SRTP protocol extension specification to IESG (Standards Track)|
|Done||Submit architecture or framework specification to IESG (Standards Track)||
|Done||Submit encrypted key transport specification to IESG (Standards Track)||