RFC6341]. In order to communicate most effectively, the SRC, the SRS, and any recording-aware UAs should utilize the mechanisms provided by RTP in a well-defined and predictable manner. It is the goal of this document to make the reader aware of these mechanisms and to provide recommendations and guidelines. RFC3550], is based on the periodic transmission of control packets to all participants in the RTP session, using the same distribution mechanism as the data packets. Support for RTCP is REQUIRED, per [RFC3550], and it provides, among other things, the following important functionality in relation to SIPREC: 1) Feedback on the quality of the data distribution This feedback from the receivers may be used to diagnose faults in the distribution. As such, RTCP is a well-defined and efficient mechanism for the SRS to inform the SRC, and for the SRC to inform recording-aware UAs, of issues that arise with respect to the reception of media that is to be recorded. 2) Including a persistent transport-level identifier -- the CNAME, or canonical name -- for an RTP source The synchronization source (SSRC) [RFC3550] identifier may change if a conflict is discovered or a program is restarted, in which case receivers can use the CNAME to keep track of each participant. Receivers may also use the CNAME to associate
multiple data streams from a given participant in a set of related RTP sessions -- for example, to synchronize audio and video. Synchronization of media streams is also facilitated by the NTP and RTP timestamps included in RTCP packets by data senders. RFC5124] when using encrypted RTP streams, and "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)" [RFC4585] when using non-encrypted media streams. However, as these are not requirements, some implementations may use "The Secure Real-time Transport Protocol (SRTP)" [RFC3711] and "RTP Profile for Audio and Video Conferences with Minimal Control" [RFC3551]. Therefore, it is RECOMMENDED that the SRC, SRS, and recording-aware UAs not rely entirely on RTP/SAVPF or RTP/AVPF for core functionality that may be at least partially achievable using RTP/SAVP and RTP/AVP. AVPF and SAVPF provide an improved RTCP timer model that allows more flexible transmission of RTCP packets in response to events, rather than strictly according to bandwidth. AVPF-based codec control messages provide efficient mechanisms for an SRC, an SRS, and recording-aware UAs to handle events such as scene changes, error recovery, and dynamic bandwidth adjustments. These messages are discussed in more detail later in this document. SAVP and SAVPF provide media encryption, integrity protection, replay protection, and a limited form of source authentication. They do not contain or require a specific keying mechanism. RFC3550], is carried in the RTP header and in various fields of RTCP packets. It is a random 32-bit number that is required to be globally unique within an RTP session. It is crucial that the number be chosen with care, in order that participants on the same network or starting at the same time are not likely to choose the same number. Guidelines regarding SSRC value selection and conflict resolution are provided in [RFC3550]. The SSRC may also be used to separate different sources of media within a single RTP session. For this reason, as well as for conflict resolution, it is important that the SRC, SRS, and recording-aware UAs handle changes in SSRC values and properly identify the reason for the change. The CNAME values carried in RTCP facilitate this identification.
RFC3550], identifies the source of a stream of RTP packets that has contributed to the combined stream produced by an RTP mixer. The mixer inserts a list of the SSRC identifiers of the sources that contributed to the generation of a particular packet into the RTP header of that packet. This list is called the CSRC list. It is RECOMMENDED that an SRC or recording-aware UA, when acting as a mixer, set the CSRC list accordingly, and that the SRC and SRS interpret the CSRC list per [RFC3550] when received. RFC3550], contains an SSRC/CSRC identifier followed by a list of zero or more items that carry information about the SSRC/CSRC. End systems send one SDES packet containing their own source identifier (the same as the SSRC in the fixed RTP header). A mixer sends one SDES packet containing a chunk for each CSRC from which it is receiving SDES information, or multiple complete SDES packets if there are more than 31 such sources. The ability to identify individual CSRCs is important in the context of SIPREC. Metadata [RFC7865] provides a mechanism to achieve this at the signaling level. SDES provides a mechanism at the RTP level. RFC3550], provides the binding from the SSRC identifier to an identifier for the source (sender or receiver) that remains constant. It is important that the SRC and recording-aware UAs generate CNAMEs appropriately and that the SRC and SRS interpret and use them for this purpose. Guidelines for generating CNAME values are provided in "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)" [RFC7022]. RFC6263] for all RTP media streams.
RFC5104] specifies extensions to the messages defined in AVPF [RFC4585]. Support for and proper usage of these messages are important to SRC, SRS, and recording-aware UA implementations. Note that these messages are applicable only when using the AVPF or SAVPF RTP profiles. RFC5168] defines an Extensible Markup Language (XML) Schema for video fast update. Implementations are discouraged from using the method described in [RFC5168], except for purposes of backward compatibility. Implementations SHOULD use FIR messages instead. To make sure that a common mechanism exists between the SRC and SRS, the SRS MUST support both mechanisms (FIR and SIP INFO), using FIR messages when negotiated successfully with the SRC and using SIP INFO otherwise. RFC4585], informs the encoder of the loss of an undefined amount of coded video data belonging to one or more pictures. [RFC4585] recommends using PLI instead of FIR messages to recover from errors. FIR is appropriate only in situations where not sending a decoder refresh point would render the video unusable for the users. Examples where sending FIR messages is appropriate include a multipoint conference when a new
user joins the conference and no regular decoder refresh point interval is established, and a video-switching Multipoint Control Unit (MCU) that changes streams. Appropriate use of PLI and FIR is important to ensure, with minimum overhead, that the recorded video is usable (e.g., the necessary reference frames exist for a player to render the recorded video). RFC5104] to request a sender to limit the maximum bit rate for a media stream to the provided value. Appropriate use of TMMBR facilitates rapid adaptation to changes in available bandwidth. RFC4961]. Likewise, when the transport address used to send and receive RTCP is the same, it is termed "symmetric RTCP" [RFC4961]. When sending RTP, the use of symmetric RTP is REQUIRED. When sending RTCP, the use of symmetric RTCP is REQUIRED. Although an SRS will not normally send RTP, it will send RTCP as well as receive RTP and RTCP. Likewise, although an SRC will not normally receive RTP from the SRS, it will receive RTCP as well as send RTP and RTCP. Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP multiplexing [RFC5761].
(Recording Session) +---------+ +------------SIP------->| | | +------RTP/RTCP----->| SRS | | | +-- Metadata -->| | | | | +---------+ v v | +---------+ | SRC | |---------| (Communication Session) +---------+ | |<----------SIP---------->| | | UA-A | | UA-B | | |<-------RTP/RTCP-------->| | +---------+ +---------+ Figure 8: UA as SRC (Recording Session) +---------+ +------------SIP------->| | | +------RTP/RTCP----->| SRS | | | +-- Metadata -->| | | | | +---------+ v v | +---------+ | SRC | +---------+ |---------| +---------+ | |<----SIP----->| |<----SIP----->| | | UA-A | | B2BUA | | UA-B | | |<--RTP/RTCP-->| |<--RTP/RTCP-->| | +---------+ +---------+ +---------+ |_______________________________________________| (Communication Session) Figure 9: B2BUA as SRC The following subsections define a set of roles an SRC may choose to play, based on its position with respect to a UA within a CS, and an SRS within an RS. A CS and a corresponding RS are independent sessions; therefore, an SRC may play a different role within a CS than it does within the corresponding RS.
RFC3550]. A defining characteristic of a translator is that it forwards RTP packets with their SSRC identifier intact. There are two types of translators: one that simply forwards, and another that performs transcoding (e.g., from one codec to another) in addition to forwarding. Section 12 provides additional information on SRTP and keying mechanisms. If packet loss occurs, either from the UA to the SRC or from the SRC to the SRS, the SRS SHOULD detect and attempt to recover from the loss. The SRC does not play a role in this, other than forwarding the associated RTP and RTCP packets.
RTCP Sender Reports generated by a UA sending a stream MUST be forwarded to the SRS. RTCP Receiver Reports generated by the SRS MUST be forwarded to the relevant UA. The SRC may need to manipulate the RTCP Receiver Reports to take into account any transcoding that has taken place. UAs may receive multiple sets of RTCP Receiver Reports -- one or more from other UAs participating in the CS, and one from the SRS participating in the RS. A recording-aware UA SHOULD be prepared to process the RTCP Receiver Reports from the SRS, whereas a recording- unaware UA may discard such RTCP packets as irrelevant. If SRTP is used on both the CS and the RS, decryption and/or re-encryption may occur. For example, if different keys are used, it will occur. If the same keys are used, it need not occur. Section 12 provides additional information on SRTP and keying mechanisms. If packet loss occurs, either from the UA to the SRC or from the SRC to the SRS, the SRS SHOULD detect and attempt to recover from the loss. The SRC does not play a role in this, other than forwarding the associated RTP and RTCP packets. RFC3550], the SRC combines RTP streams from different UAs and sends them towards the SRS using its own SSRC. The SSRCs from the contributing UA SHOULD be conveyed as CSRC identifiers within this stream. The SRC may make timing adjustments among the received streams and generate its own timing on the stream sent to the SRS. Optionally, an SRC acting as a mixer can perform transcoding and can even cope with different codings received from different UAs. RTCP Sender Reports and Receiver Reports are not forwarded by an SRC acting as a mixer, but there are requirements for forwarding RTCP Source Description (SDES) packets. The SRC generates its own RTCP Sender Reports and Receiver Reports toward the associated UAs and SRS. The use of SRTP between the SRC and the SRS for the RS is independent of the use of SRTP between the UAs and the SRC for the CS. Section 12 provides additional information on SRTP and keying mechanisms. If packet loss occurs from the UA to the SRC, the SRC SHOULD detect and attempt to recover from the loss. If packet loss occurs from the SRC to the SRS, the SRS SHOULD detect and attempt to recover from the loss.
RFC3550], is similar to the mixer case, except that the RTP session between the SRC and the SRS is considered completely independent from the RTP session that is part of the CS. The SRC can, but need not, mix RTP streams from different participants prior to sending to the SRS. RTCP between the SRC and the SRS is completely independent of RTCP on the CS. The use of SRTP between the SRC and the SRS for the RS is independent of the use of SRTP between the UAs and SRC for the CS. Section 12 provides additional information on SRTP and keying mechanisms. If packet loss occurs from the UA to the SRC, the SRC SHOULD detect and attempt to recover from the loss. If packet loss occurs from the SRC to the SRS, the SRS SHOULD detect and attempt to recover from the loss. RFC3264]. Having received the answer, the SRC starts sending media to the SRS as indicated in the answer. Alternatively, if the SRC deems the level of support indicated in the answer to be unacceptable, it may initiate another SDP offer/answer exchange in which an alternative RTP session usage is negotiated.
In order to preserve the mapping of media to participant within the CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to a unique CNAME within the RS. Additionally, the SRC SHOULD map each unique combination of CNAME/SSRC within the CSs to a unique CNAME/SSRC within the RS. In doing so, the SRC may act as an RTP translator or as an RTP endpoint. Figure 10 illustrates a case in which each UA represents a participant contributing two RTP sessions (e.g., one for audio and one for video), each with a single SSRC. The SRC acts as an RTP translator and delivers the media to the SRS using four RTP sessions, each with a single SSRC. The CNAME and SSRC values used by the UAs within their media streams are preserved in the media streams from the SRC to the SRS. +---------+ +------------SSRC Aa--->| | | + --------SSRC Av--->| | | | +------SSRC Ba--->| SRS | | | | +---SSRC Bv--->| | | | | | +---------+ | | | | | | | | +---------+ +----------+ +---------+ | |---SSRC Aa-->| SRC |<--SSRC Ba---| | | UA-A | |(CNAME-A, | | UA-B | |(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)| +---------+ +----------+ +---------+ Figure 10: SRC Using Multiple m-lines RFC3264]. Having received the answer, the SRC starts sending media to the SRS as indicated in the answer. In order to preserve the mapping of media to participant within the CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to a unique CNAME within the RS. Additionally, the SRC SHOULD map each unique combination of CNAME/SSRC within the CSs to a unique
CNAME/SSRC within the RS. The SRC MUST avoid SSRC collisions, rewriting SSRCs if necessary when used as CSRCs in the RS. In doing so, the SRC acts as an RTP mixer. In the event that the SRS does not support this usage of CSRC values, it relies entirely on the SIPREC metadata to determine the participants included within each mixed stream. Figure 11 illustrates a case in which each UA represents a participant contributing two RTP sessions (e.g., one for audio and one for video), each with a single SSRC. The SRC acts as an RTP mixer and delivers the media to the SRS using two RTP sessions, mixing media from each participant into a single RTP session containing a single SSRC and two CSRCs. SSRC Sa +---------+ +-------CSRC Aa,Ba--->| | | | | | SSRC Sv | SRS | | +---CSRC Av,Bv--->| | | | +---------+ | | +----------+ +---------+ | SRC | +---------+ | |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---| | | UA-A | | CNAME-A, | | UA-B | |(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)| +---------+ +----------+ +---------+ Figure 11: SRC Using Mixing
other modes from being implemented and used, such as using a single m-line and mixing the separate audio streams together. Rather, the requirements were written to provide a common base mode to implement for the sake of interoperability. It is important to note that an SRS implementation supporting the common base mode may not record all media streams in a CS if a participant supports more than one m-line in a video call, such as one for camera and one for presentation. SRS implementations may support other modes as well, but they have to at least support the modes discussed above, such that they interoperate in the common base mode for basic interoperability. RFC7865]. A new "disposition-type" of Content-Disposition is defined for the purpose of carrying metadata. The value is "recording-session", which indicates that the "application/rs-metadata" content contains metadata to be handled by the SRS.
The SRC MAY send metadata (either a full metadata snapshot or a partial update) in an INVITE request, an UPDATE request [RFC3311], or a 200 response to an offerless INVITE from the SRS. If the metadata contains a reference to any SDP labels, the request containing the metadata MUST also contain an SDP offer that defines those labels. When a SIP message contains both an SDP offer and metadata, the request body MUST have content type "multipart/mixed", with one subordinate body part containing the SDP offer and another containing the metadata. When a SIP message contains only an SDP offer or metadata, the "multipart/mixed" container is optional. The SRC SHOULD include a full metadata snapshot in the initial INVITE request establishing the RS. If metadata is not yet available (e.g., an RS established in the absence of a CS), the SRC SHOULD send a full metadata snapshot as soon as metadata becomes available. If the SRC receives a snapshot request from the SRS, it MUST immediately send a full metadata snapshot.
Figure 12 illustrates an example of a full metadata snapshot sent by the SRC in the initial INVITE request: INVITE sip:firstname.lastname@example.org SIP/2.0 Via: SIP/2.0/TCP src.example.com;branch=z9hG4bKdf6b622b648d9 From: <sip:email@example.com>;tag=35e195d2-947d-4585-946f-09839247 To: <sip:firstname.lastname@example.org> Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a CSeq: 101 INVITE Max-Forwards: 70 Require: siprec Accept: application/sdp, application/rs-metadata Contact: <sip:email@example.com>;+sip.src Content-Type: multipart/mixed;boundary=foobar Content-Length: [length] --foobar Content-Type: application/sdp v=0 o=SRS 2890844526 2890844526 IN IP4 198.51.100.1 s=- c=IN IP4 198.51.100.1 t=0 0 m=audio 12240 RTP/AVP 0 4 8 a=sendonly a=label:1 --foobar Content-Type: application/rs-metadata Content-Disposition: recording-session [metadata content] Figure 12: Sample INVITE Request for the Recording Session
error on the SRC side, and it is likely that the same error will occur again even when a full metadata snapshot is requested. In order to avoid repeating the same error, the SRS can simply terminate the RS when a syntax error or semantic error is detected in the metadata. The SRS MAY explicitly request a full metadata snapshot by sending an UPDATE request. This request MUST contain a body with Content-Disposition type "recording-session" and MUST NOT contain an SDP body. The SRS MUST NOT request a full metadata snapshot in an UPDATE response or in any other SIP transaction. The format of the content is "application/rs-metadata", and the body is an XML document, the format of which is defined in [RFC7865]. Figure 13 shows an example: UPDATE sip:firstname.lastname@example.org SIP/2.0 Via: SIP/2.0/UDP srs.example.com;branch=z9hG4bKdf6b622b648d9 To: <sip:email@example.com>;tag=35e195d2-947d-4585-946f-098392474 From: <sip:firstname.lastname@example.org>;tag=1234567890 Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a CSeq: 1 UPDATE Max-Forwards: 70 Require: siprec Contact: <sip:email@example.com>;+sip.srs Accept: application/sdp, application/rs-metadata Content-Disposition: recording-session Content-Type: application/rs-metadata Content-Length: [length] <?xml version="1.0" encoding="UTF-8"?> <requestsnapshot xmlns='urn:ietf:params:xml:ns:recording:1'> <requestreason xml:lang="it">SRS internal error</requestreason> </requestsnapshot> Figure 13: Metadata Request Note that UPDATE was chosen for the SRS to request a metadata snapshot, because it can be sent regardless of the state of the dialog. This was seen as better than requiring support for both UPDATE and re-INVITE messages for this operation. When the SRC receives a request for a metadata snapshot, it MUST immediately provide a full metadata snapshot in a separate INVITE or UPDATE transaction. Any subsequent partial updates will not be dependent on any metadata sent prior to this full metadata snapshot.
The metadata received by the SRS can contain ID elements used to cross-reference one element to another. An element containing the definition of an ID and an element containing a reference to that ID will often be received from the same SRC. It is also valid for those elements to be received from different SRCs -- for example, when each endpoint in the same CS acts as an SRC to record the call and a common ID refers to the same CS. The SRS MUST NOT consider this an error. RFC6341], where an RS can be established in the absence of a CS. The SRC continuously records media in an RS to the SRS even in the absence of a CS for all UAs that are part of persistent recording. By allocating recorded streams and continuously sending recorded media to the SRS, the SRC does not have to prepare new recorded streams with a new SDP offer when a new CS is created and also does not impact the timing of the CS. The SRC only needs to update the metadata when new CSs are created. When there is no CS running on the devices with persistent recording, there is no recorded media to stream from the SRC to the SRS. In certain environments where a Network Address Translator (NAT) is used, a minimum amount of flow activity is typically required to maintain the NAT binding for each port opened. Agents that support Interactive Connectivity Establishment (ICE) solve this problem. For non-ICE agents, in order not to lose the NAT bindings for the RTP/RTCP ports opened for the recorded streams, the SRC and SRS SHOULD follow the recommendations provided in [RFC6263] to maintain the NAT bindings.
Examples of typical use: Routing the request to a Session Recording Server. Security Considerations: Security considerations for this media feature tag are discussed in Section 11.1 of RFC 3840. Section 11.1 of RFC 3840.
RFC3261]; therefore, the RS can reuse any of the existing SIP security mechanisms available for securing the session signaling, the recorded media, and the metadata. The use cases and requirements document [RFC6341] outlines the general security considerations, and this document describes specific security recommendations. The SRC and SRS MUST support SIP with Transport Layer Security (TLS) version 1.2, SHOULD follow the best practices when using TLS as per [RFC7525], and MAY use Session Initiation Protocol Secure (SIPS) with TLS as per [RFC5630]. The RS MUST be at least as secure as the CS; this means using at least the same strength of cipher suite as the CS if the CS is secured. For example, if the CS uses SIPS for signaling and RTP/SAVP for media, then the RS may not use SIP or plain RTP unless other equivalent security measures are in effect, since doing so would mean an effective security downgrade. Examples of other potentially equivalent security mechanisms include mutually authenticated TLS for the RS signaling channel or an appropriately protected network path for the RS media component.
utilized. Additionally, an SRC or SRS may use other existing SIP mechanisms available, including, but not limited to, Digest authentication [RFC3261], asserted identity [RFC3325], and connected identity [RFC4916]. The SRS may have its own set of recording policies to authorize recording requests from the SRC. The use of recording policies is outside the scope of the Session Recording Protocol. RFC3711] and RTP/SAVPF [RFC5124]. RTP/SAVP and RTP/SAVPF provide media encryption, integrity protection, replay protection, and a limited form of source authentication. They do not contain or require a specific keying mechanism. At a minimum, the SRC and SRS MUST support the SDP security descriptions key negotiation mechanism [RFC4568]. For cases in which Datagram Transport Layer Security for Secure RTP (DTLS-SRTP) is used to encrypt a CS media stream, an SRC may use SRTP Encrypted Key Transport (EKT) [EKT-SRTP] in order to use SRTP-SDES in the RS without needing to re-encrypt the media. Note: When using EKT in this manner, it is possible for participants in the CS to send traffic that appears to be from other participants and have this forwarded by the SRC to the SRS within the RS. If this is a concern (e.g., the RS is intended for audit or compliance purposes), EKT is not an appropriate choice. When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same keys or different keys in the RS than those used in the CS. Some SRCs are designed to simply replicate RTP packets from a CS media stream to the SRS, in which case the SRC will use the same key in the RS as the key used in the CS. In this case, the SRC MUST secure the SDP containing the keying material in the RS with at least the same level of security as in the CS. The risk of lowering the level of security in the RS is that it will effectively become a downgrade attack on the CS, since the same key is used for both the CS and the RS. SRCs that decrypt an encrypted CS media stream and re-encrypt it when sending it to the SRS MUST use a different key than what is used for the CS media stream, to ensure that it is not possible for someone who has the key for the CS media stream to access recorded data they
are not authorized to access. In order to maintain a comparable level of security, the key used in the RS SHOULD be of equivalent strength to, or greater strength than, that used in the CS. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <http://www.rfc-editor.org/info/rfc2119>. [RFC2506] Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag Registration Procedure", BCP 31, RFC 2506, DOI 10.17487/RFC2506, March 1999, <http://www.rfc-editor.org/info/rfc2506>. [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, DOI 10.17487/RFC3261, June 2002, <http://www.rfc-editor.org/info/rfc3261>. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, DOI 10.17487/RFC3264, June 2002, <http://www.rfc-editor.org/info/rfc3264>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, July 2003, <http://www.rfc-editor.org/info/rfc3550>. [RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)", RFC 3840, DOI 10.17487/RFC3840, August 2004, <http://www.rfc-editor.org/info/rfc3840>. [RFC4574] Levin, O. and G. Camarillo, "The Session Description Protocol (SDP) Label Attribute", RFC 4574, DOI 10.17487/RFC4574, August 2006, <http://www.rfc-editor.org/info/rfc4574>. [RFC5234] Crocker, D., Ed., and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, DOI 10.17487/RFC5234, January 2008, <http://www.rfc-editor.org/info/rfc5234>. [RFC7245] Hutton, A., Ed., Portman, L., Ed., Jain, R., and K. Rehor, "An Architecture for Media Recording Using the Session Initiation Protocol", RFC 7245, DOI 10.17487/RFC7245, May 2014, <http://www.rfc-editor.org/info/rfc7245>. [RFC7865] Ravindranath, R., Ravindran, P., and P. Kyzivat, "Session Initiation Protocol (SIP) Recording Metadata", RFC 7865, DOI 10.17487/RFC7865, May 2016, <http://www.rfc-editor.org/info/rfc7865>. [EKT-SRTP] Mattsson, J., Ed., McGrew, D., Wing, D., and F. Andreasen, "Encrypted Key Transport for Secure RTP", Work in Progress, draft-ietf-avtcore-srtp-ekt-03, October 2014. [RFC2804] IAB and IESG, "IETF Policy on Wiretapping", RFC 2804, DOI 10.17487/RFC2804, May 2000, <http://www.rfc-editor.org/info/rfc2804>. [RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE Method", RFC 3311, DOI 10.17487/RFC3311, October 2002, <http://www.rfc-editor.org/info/rfc3311>.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", RFC 3325, DOI 10.17487/RFC3325, November 2002, <http://www.rfc-editor.org/info/rfc3325>. [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, DOI 10.17487/RFC3551, July 2003, <http://www.rfc-editor.org/info/rfc3551>. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, DOI 10.17487/RFC3711, March 2004, <http://www.rfc-editor.org/info/rfc3711>. [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006, <http://www.rfc-editor.org/info/rfc4568>. [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, DOI 10.17487/RFC4585, July 2006, <http://www.rfc-editor.org/info/rfc4585>. [RFC4916] Elwell, J., "Connected Identity in the Session Initiation Protocol (SIP)", RFC 4916, DOI 10.17487/RFC4916, June 2007, <http://www.rfc-editor.org/info/rfc4916>. [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007, <http://www.rfc-editor.org/info/rfc4961>. [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, February 2008, <http://www.rfc-editor.org/info/rfc5104>. [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February 2008, <http://www.rfc-editor.org/info/rfc5124>. [RFC5168] Levin, O., Even, R., and P. Hagendorf, "XML Schema for Media Control", RFC 5168, DOI 10.17487/RFC5168, March 2008, <http://www.rfc-editor.org/info/rfc5168>.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)", RFC 5630, DOI 10.17487/RFC5630, October 2009, <http://www.rfc-editor.org/info/rfc5630>. [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", RFC 5761, DOI 10.17487/RFC5761, April 2010, <http://www.rfc-editor.org/info/rfc5761>. [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows", RFC 6263, DOI 10.17487/RFC6263, June 2011, <http://www.rfc-editor.org/info/rfc6263>. [RFC6341] Rehor, K., Ed., Portman, L., Ed., Hutton, A., and R. Jain, "Use Cases and Requirements for SIP-Based Media Recording (SIPREC)", RFC 6341, DOI 10.17487/RFC6341, August 2011, <http://www.rfc-editor.org/info/rfc6341>. [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, September 2013, <http://www.rfc-editor.org/info/rfc7022>. [RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre, "Recommendations for Secure Use of Transport Layer Security (TLS) and Datagram Transport Layer Security (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May 2015, <http://www.rfc-editor.org/info/rfc7525>.