Internet Engineering Task Force (IETF) J. Xia Request for Comments: 8286 R. Even Category: Standards Track R. Huang ISSN: 2070-1721 Huawei L. Deng China Mobile October 2017 RTP/RTCP Extension for RTP Splicing Notification
AbstractContent splicing is a process that replaces the content of a main multimedia stream with other multimedia content and that delivers the substitutive multimedia content to the receivers for a period of time. The splicer is designed to handle RTP splicing and needs to know when to start and end the splicing. This memo defines two RTP/RTCP extensions to indicate the splicing- related information to the splicer: an RTP header extension that conveys the information "in band" and an RTP Control Protocol (RTCP) packet that conveys the information out of band. Status of This Memo This is an Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at https://www.rfc-editor.org/info/rfc8286.
Copyright Notice Copyright (c) 2017 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. 1. Introduction ....................................................3 1.1. Terminology ................................................3 2. Overview ........................................................4 2.1. Overview of RTP Splicing ...................................4 2.2. Overview of Splicing Interval ..............................5 3. Conveying Splicing Interval in RTP/RTCP Extensions ..............7 3.1. RTP Header Extension .......................................7 3.2. RTCP Splicing Notification Message .........................8 4. Reducing Splicing Latency ......................................10 5. Failure Cases ..................................................11 6. Session Description Protocol (SDP) Signaling ...................12 6.1. Declarative SDP ...........................................12 6.2. Offer/Answer without BUNDLE ...............................13 6.3. Offer/Answer with BUNDLE: All Media Are Spliced ...........14 6.4. Offer/Answer with BUNDLE: A Subset of Media Are Spliced ...16 7. Security Considerations ........................................18 8. IANA Considerations ............................................19 8.1. RTCP Control Packet Types .................................19 8.2. RTP Compact Header Extensions .............................20 8.3. SDP Grouping Semantic Extension ...........................20 9. References .....................................................20 9.1. Normative References ......................................20 9.2. Informative References ....................................21 Acknowledgements ..................................................22 Authors' Addresses ................................................22
SCTE35] provides a method that encapsulates the Splicing Interval inside the MPEG2-TS (MPEG2 transport stream) layer in cable TV systems. When transported in RTP, a middlebox designed as the splicer to decode the RTP packets and search for the Splicing Interval inside the payloads is required. The need for such processing increases the workload of the middlebox and limits the number of RTP sessions the middlebox can support. This document defines an RTP header extension [RFC8285] used by the main RTP sender to provide the Splicing Interval by including it in the RTP packets. However, the Splicing Interval conveyed in the RTP header extension might not reach the splicer successfully. Any splicing-unaware middlebox on the path between the RTP sender and the splicer might strip this RTP header extension. To increase robustness against such a case, this document also defines a new RTP Control Protocol (RTCP) packet type to carry the same Splicing Interval to the splicer. Since RTCP is also unreliable and may not be as "immediate" as the in-band technique, it's only considered to be a complement to the RTP header extension. RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
In addition, we define the following terms: Main RTP Sender: The sender of RTP packets carrying the main RTP stream. Splicer: An intermediary node that inserts substitutive content into a main RTP stream. The splicer sends substitutive content to the RTP receiver instead of the main content during splicing. It is also responsible for processing RTCP traffic between the RTP sender and the RTP receiver. Splicing-In Point: A virtual point in the RTP stream, suitable for substitutive content entry, typically in the boundary between two independently decodable frames. Splicing-Out Point: A virtual point in the RTP stream, suitable for substitutive content exit, typically in the boundary between two independently decodable frames. Splicing Interval: The NTP timestamps, representing the main RTP sender wallclock time, for the splicing-in point and splicing-out point per [RFC6828], allowing the splicer to know when to start and end the RTP splicing. Substitutive RTP Sender: The sender of RTP packets carrying the RTP stream that will replace the content in the main RTP stream.
scenario is where an IPTV service provider uses its own regional advertising content to replace national advertising content, the time window of which is explicitly indicated by the IPTV service provider. The splicer is a middlebox handling RTP splicing. It receives the main content and substitutive content simultaneously but only chooses to send one of them to the receiver at any point in time. When RTP splicing begins, the splicer sends the substitutive content to the receivers instead of the main content. When RTP splicing ends, the splicer switches back to sending the main content to the receivers. This implies that the receiver is explicitly configured to receive the traffic via the splicer and will return any RTCP feedback to it in the presence of the splicer. The middlebox working as the splicer can be implemented as either an RTP mixer or an RTP translator. If implemented as an RTP mixer, the splicer will use its own synchronization source (SSRC), sequence number space, and timing model when generating the output stream to receivers, using the contributing source (CSRC) list to indicate whether the original content or substitutive content is being delivered. The splicer, on behalf of the content provider, can omit the CSRC list from the RTP packets it generates. This simplifies the design of the receivers, since they don't need to parse the CSRC list, but makes it harder to determine when the splicing is taking place (it requires inspection of the RTP payload data, rather than just the RTP headers). A splicer working as an RTP mixer splits the flow between the sender and receiver into two, and it requires separate control loops for RTCP and congestion control. [RFC6828] provides an example of an RTP mixer approach. A splicer implemented as an RTP translator [RFC3550] will forward the RTP packets from the original and substitutive senders with their SSRCs intact but will need to rewrite RTCP Sender Report (SR) packets to account for the splicing. In this case, the congestion control loops run between the original sender and receiver and between the substitutive sender and receiver. The splicer needs to ensure that the RTCP feedback messages from the receiver are passed to the right sender to let the congestion control work.
than once to mitigate possible packet loss. To enable the splicer to get the substitutive content before the splicing starts, the main RTP sender MUST send the Splicing Interval well in advance. For example, the main RTP sender can estimate when to send the Splicing Interval based on the round-trip time (RTT), following the mechanisms described in Section 6.4.1 of [RFC3550] when the splicer sends an RTCP Receiver Report (RR) to the main sender. The substitutive sender also needs to learn the Splicing Interval from the main RTP sender in advance and estimate when to transfer the substitutive content to the splicer. The Splicing Interval could be transmitted from the main RTP sender to the substitutive content using some out-of-band mechanisms -- for example, a proprietary mechanism to exchange the Splicing Interval -- or the substitutive sender is implemented together with the main RTP sender inside a single device. To ensure that the Splicing Interval is valid for both the main RTP sender and the substitutive RTP sender, the two senders MUST share a common reference clock so that the splicer can achieve accurate splicing. The requirements for the common reference clock (e.g., resolution, skew) depend on the codec used by the media content. In this document, the main RTP sender uses a pair of NTP timestamps to indicate when to start and end the splicing to the splicer: the timestamp of the first substitutive RTP packet at the splicing-in point and the timestamp of the first main RTP packet at the splicing-out point. When the substitutive RTP sender gets the Splicing Interval, it must prepare the substitutive stream. The main content provider and the substitutive content provider MUST ensure that the RTP timestamp of the first substitutive RTP packet that would be presented to the receivers corresponds to the same time instant as the former NTP timestamp in the Splicing Interval. To enable the splicer to know the first substitutive RTP packet it needs to send, the substitutive RTP sender MUST send the substitutive RTP packet ahead of the splicing-in point, allowing the splicer to find out the timestamp of this first RTP packet in the substitutive RTP stream, e.g., using a prior RTCP SR message. When it is time for the splicing to end, the main content provider and the substitutive content provider MUST ensure that the RTP timestamp of the first main RTP packet that would be presented on the receivers corresponds to the same time instant as the latter NTP timestamp in the Splicing Interval.
RFC8285] can be adapted to carry the Splicing Interval, which consists of a pair of NTP timestamps. This RTP header extension carries the 7 octets of the splicing-out NTP timestamp (lower 24-bit part of the "Seconds" of an NTP timestamp and the 32 bits of the "Fraction" of an NTP timestamp as defined in [RFC5905]), followed by the 8 octets of the splicing-in NTP timestamp (64-bit NTP timestamp as defined in [RFC5905]). The top 8 bits of the splicing-out NTP timestamp are inferred from the top 8 bits of the splicing-in NTP timestamp, assuming that (1) the splicing-out time is after the splicing-in time and (2) the Splicing Interval is less than 2^25 seconds. Therefore, if the value of the 7 octets of the splicing-out NTP timestamp is smaller than the value of the 7 lower octets of the splicing-in NTP timestamp, it implies a wrap of the 56-bit splicing-out NTP timestamp, which means that the top 8-bit value of the 64-bit splicing-out NTP timestamp is equal to the top 8-bit value of the splicing-in NTP timestamp plus 0x01. Otherwise, the top 8 bits of the splicing-out NTP timestamp are equal to the top 8 bits of the splicing-in NTP timestamp. This RTP header extension can be encoded using either the one-byte or two-byte header defined in [RFC8285]. Figures 1 and 2 show the Splicing Interval header extension with each of the two header formats.
0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E | ID | L=14 | OUT NTP timestamp - Seconds (bit 8-31) |x +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t | OUT NTP timestamp - Fraction (bit 0-31) |e +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n | IN NTP timestamp - Seconds (bit 0-31) |s +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i | IN NTP timestamp - Fraction (bit 0-31) |o +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n Figure 1: Splicing Interval Using the One-Byte Header Format 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+E | ID | L=15 | OUT NTP timestamp - Seconds |x +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+t |OUT Secds(cont)| OUT NTP timestamp - Fraction |e +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n |OUT Fract(cont)| IN NTP timestamp - Seconds |s +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+i | IN Secds(cont)| IN NTP timestamp - Fraction |o +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+n | IN Fract(cont)| 0 (pad) | ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 2: Splicing Interval Using the Two-Byte Header Format Since the inclusion of an RTP header extension will reduce the efficiency of RTP header compression, it is RECOMMENDED that the main sender insert the RTP header extensions into a number of RTP packets, instead of all of the RTP packets, prior to the splicing-in. After the splicer obtains the RTP header extension and derives the Splicing Interval, it generates its own stream and is not allowed to include the RTP header extension in outgoing packets; this reduces header overhead.
MUST make sure that the splicing information contained in the RTCP splicing notification message is consistent with the information included in the RTP header extensions. The RTCP splicing notification message is a new RTCP packet type. It has a fixed header followed by a pair of NTP timestamps: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|reserved | PT=213 | length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | IN NTP timestamp (most significant word) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | IN NTP timestamp (least significant word) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | OUT NTP timestamp (most significant word) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | OUT NTP timestamp (least significant word) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 3: RTCP Splicing Notification Message The RTCP splicing notification message includes the following fields: Length: 16 bits As defined in [RFC3550], the length of the RTCP packet in 32-bit words minus one, including the header and any padding. SSRC: 32 bits The SSRC of the main RTP sender. Timestamp: 64 bits Indicates the wallclock time when this splicing starts and ends. The full-resolution NTP timestamp is used, which is a 64-bit unsigned fixed-point number with the integer part in the first 32 bits and the fractional part in the last 32 bits. This format is the same as the NTP timestamp field in the RTCP SR (Section 6.4.1 of [RFC3550]).
The RTCP splicing notification message can be included in the RTCP compound packet together with the RTCP SR generated at the main RTP sender; hence, it follows the compound RTCP rules defined in Section 6.1 in [RFC3550]. If the use of non-compound RTCP [RFC5506] was previously negotiated between the sender and the splicer, the RTCP splicing notification messages may be sent as non-compound RTCP packets. In some cases where the mapping from the RTP timestamp to the NTP timestamp changes, e.g., clock drift happens before the splicing event, sending an RTCP SR or even updated Splicing Interval information in a timely manner might be required in order to update the timestamp mapping for accurate splicing. Since the RTCP splicing notification message is intentionally sent by the main RTP sender to the splicer, the splicer is not allowed to forward this message to the receivers, so as to avoid useless processing and additional RTCP bandwidth consumption in the downstream receivers. RFC6285] refers to this information as "Reference Information") to start processing the multimedia data, either the main RTP sender or the substitutive sender SHOULD provide the Reference Information together with its multimedia content to reduce the delay caused by acquiring the Reference Information. The methods by which the Reference Information is distributed to the receivers are out of scope for this memo. Another latency element is delay caused by synchronization. The receivers must receive enough synchronization metadata prior to synchronizing the separate components of the multimedia streams when splicing starts or ends. Either the main RTP sender or the substitutive sender SHOULD send the synchronization metadata early enough so that the receivers can play out the multimedia in a synchronized fashion. The main RTP sender or the substitutive sender can estimate when to send the synchronization metadata based on, for example, the RTT, following the mechanisms described in Section 6.4.1 of [RFC3550] when the splicer sends an RTCP RR to the main sender or the substitutive sender. The main RTP sender and the substitutive sender can also be coordinated by some proprietary out-of-band mechanisms to decide when, and to whom, the metadata is to be sent. If both send the information, the splicer SHOULD pick one based on the current situation, e.g., choosing either (1) the main RTP sender
RFC5888] with a new semantic, called "SPLICE", to represent the relationship between the main RTP stream and the substitutive RTP stream. Only two "m=" lines are allowed in the SPLICE group. The main RTP stream is the one with the extended "extmap" attribute, and the other one is the substitutive stream. A single "m=" line MUST NOT be included in different SPLICE groups at the same time. The main RTP sender provides the information about both main and substitutive sources. The extended SDP attribute specified in this document is applicable for offer/answer content [RFC3264] and does not affect any rules when negotiating offers and answers. When used with multiple "m=" lines, substitutive RTP MUST be applied only to the RTP packets whose SDP "m=" line is in the same group with the substitutive stream using SPLICE and has the extended splicing "extmap" attribute. This semantic is also applicable for BUNDLE cases. The following examples show how SDP signaling could be used for splicing in different cases. v=0 o=xia 1122334455 1122334466 IN IP4 splicing.example.com s=RTP Splicing Example t=0 0 a=group:SPLICE 1 2 m=video 30000 RTP/AVP 100 i=Main RTP Stream c=IN IP4 184.108.40.206/127 a=rtpmap:100 MP2T/90000 a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval a=mid:1 m=video 30002 RTP/AVP 100 i=Substitutive RTP Stream c=IN IP4 220.127.116.11/127 a=sendonly a=rtpmap:100 MP2T/90000 a=mid:2 Figure 4: Example SDP for a Single-Channel Splicing Scenario
The splicer receiving the SDP message above receives one MPEG2-TS stream (payload 100) from the main RTP sender (with a multicast destination address of 18.104.22.168) on port 30000 and/or receives another MPEG2-TS stream from the substitutive RTP sender (with a multicast destination address of 22.214.171.124) on port 30002. But at a particular point in time, the splicer only selects one stream and outputs the content from the chosen stream to the downstream receivers.
SDP Answer - from the splicer: v=0 o=xia 1122334455 1122334466 IN IP4 splicer.example.com s=RTP Splicing Example t=0 0 a=group:SPLICE 1 2 m=video 30000 RTP/AVP 100 i=Main RTP Stream c=IN IP4 splicer.example.com a=rtpmap:100 MP2T/90000 a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval a=recvonly a=mid:1 m=video 40000 RTP/AVP 100 i=Substitutive RTP Stream c=IN IP4 splicer.example.com a=rtpmap:100 MP2T/90000 a=recvonly a=mid:2
SDP Offer - from the main RTP sender: v=0 o=alice 1122334455 1122334466 IN IP4 splicing.example.com s=RTP Splicing Example c=IN IP4 splicing.example.com t=0 0 a=group:SPLICE foo 1 a=group:SPLICE bar 2 a=group:BUNDLE foo bar m=audio 10000 RTP/AVP 0 8 97 a=mid:foo b=AS:200 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval a=sendonly m=video 10002 RTP/AVP 31 32 a=mid:bar b=AS:1000 a=rtpmap:31 H261/90000 a=rtpmap:32 MPV/90000 a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval a=sendonly m=audio 20000 RTP/AVP 0 8 97 i=Substitutive audio RTP Stream c=IN IP4 substitutive.example.com a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=sendonly a=mid:1 m=video 20002 RTP/AVP 31 32 i=Substitutive video RTP Stream c=IN IP4 substitutive.example.com a=rtpmap:31 H261/90000 a=rtpmap:32 MPV/90000 a=mid:2 a=sendonly
SDP Answer - from the splicer: v=0 o=bob 2808844564 2808844564 IN IP4 splicer.example.com s=RTP Splicing Example c=IN IP4 splicer.example.com t=0 0 a=group:SPLICE foo 1 a=group:SPLICE bar 2 a=group:BUNDLE foo bar m=audio 30000 RTP/AVP 0 a=mid:foo b=AS:200 a=rtpmap:0 PCMU/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:splicing-interval a=recvonly m=video 30000 RTP/AVP 32 a=mid:bar b=AS:1000 a=rtpmap:32 MPV/90000 a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval a=recvonly m=audio 30002 RTP/AVP 0 i=Substitutive audio RTP Stream c=IN IP4 splicer.example.com a=rtpmap:0 PCMU/8000 a=recvonly a=mid:1 m=video 30004 RTP/AVP 32 i=Substitutive video RTP Stream c=IN IP4 splicer.example.com a=rtpmap:32 MPV/90000 a=mid:2 a=recvonly
SDP Offer - from the main RTP sender: v=0 o=alice 1122334455 1122334466 IN IP4 splicing.example.com s=RTP Splicing Example c=IN IP4 splicing.example.com t=0 0 a=group:SPLICE bar 2 a=group:BUNDLE foo bar m=audio 10000 RTP/AVP 0 8 97 a=mid:foo b=AS:200 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=sendonly m=video 10002 RTP/AVP 31 32 a=mid:bar b=AS:1000 a=rtpmap:31 H261/90000 a=rtpmap:32 MPV/90000 a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval a=sendonly m=video 20000 RTP/AVP 31 32 i=Substitutive video RTP Stream c=IN IP4 substitutive.example.com a=rtpmap:31 H261/90000 a=rtpmap:32 MPV/90000 a=mid:2 a=sendonly
SDP Answer - from the splicer: v=0 o=bob 2808844564 2808844564 IN IP4 splicer.example.com s=RTP Splicing Example c=IN IP4 splicer.example.com t=0 0 a=group:SPLICE bar 2 a=group:BUNDLE foo bar m=audio 30000 RTP/AVP 0 a=mid:foo b=AS:200 a=rtpmap:0 PCMU/8000 a=recvonly m=video 30000 RTP/AVP 32 a=mid:bar b=AS:1000 a=rtpmap:32 MPV/90000 a=extmap:2 urn:ietf:params:rtp-hdrext:splicing-interval a=recvonly m=video 30004 RTP/AVP 32 i=Substitutive video RTP Stream c=IN IP4 splicer.example.com a=rtpmap:32 MPV/90000 a=mid:2 a=recvonly RFC3550] and the general mechanism for RTP header extensions [RFC8285] apply. The splicer can be either a mixer or a translator, and all the security considerations of topologies [RFC7667] [RFC7201] for these two types of RTP intermediaries are applicable for the splicer. The splicer replaces some content with other content in RTP packets, thus breaking any RTP-level end-to-end security, such as source authentication and integrity protection. End-to-end source authentication is not possible with any known existing splicing solution. A new solution can theoretically be developed that enables identification of the participating entities and what each provides, i.e., the different media sources -- main and substitutive -- and the splicer, which provides the RTP-level integration of the media payloads in a common timeline and synchronization context. Since the splicer breaks RTP-level end-to-end security, it needs to be part of the signaling context and the necessary security associations (e.g., Secure Real-time Transport Protocol (SRTP)
[RFC3711] crypto contexts) established for the RTP session participants. When using SRTP, the splicer would have to be provisioned with the same security association as the main RTP sender. If there are concerns about the confidentiality of the splicing time information, the header extension defined in this document MUST also be protected; for example, header extension encryption [RFC6904] can be used in this case. However, the malicious endpoint may get the splicing time information by other means, e.g., inferring it from the communication between the main and substitutive content sources. To avoid the insertion of invalid substitutive content, the splicer MUST have some mechanisms to authenticate the substitutive stream source. For cases where the splicing time information is changed by a malicious endpoint, the splicing, for example, may fail, since it will not be available at the right time for the substitutive media to arrive. Another case is one where an attacker may prevent the receivers from receiving the content from the main sender by inserting extra splicing time information. To avoid the above scenarios, the authentication of the RTP header extension for splicing time information SHOULD be considered. When a splicer implemented as a mixer sends the stream to the receivers, the CSRC list, which can be used to detect RTP-level forwarding loops as defined in Section 8.2 of [RFC3550], may be removed for simplifying the receivers that cannot handle multiple sources in the RTP stream. Hence, loops may occur, causing packets to loop back to a point upstream of the splicer and possibly forming a serious denial-of-service threat. In such a case, non-RTP means, e.g., signaling among all the participants, MUST be used to detect and resolve loops. RFC8126], a new RTCP packet format has been registered in the "RTCP Control Packet types (PT)" registry: Name: SNM Long name: Splicing Notification Message Value: 213 Reference: This document
RFC8285], according to the following: Extension URI: urn:ietf:params:rtp-hdrext:splicing-interval Description: Splicing Interval Contact: Jinwei Xia <firstname.lastname@example.org> Reference: This document [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <https://www.rfc-editor.org/info/rfc2119>. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, DOI 10.17487/RFC3264, June 2002, <https://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, <https://www.rfc-editor.org/info/rfc3550>. [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description Protocol (SDP) Grouping Framework", RFC 5888, DOI 10.17487/RFC5888, June 2010, <https://www.rfc-editor.org/info/rfc5888>.
[RFC5905] Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch, "Network Time Protocol Version 4: Protocol and Algorithms Specification", RFC 5905, DOI 10.17487/RFC5905, June 2010, <https://www.rfc-editor.org/info/rfc5905>. [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010, <https://www.rfc-editor.org/info/rfc6051>. [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014, <https://www.rfc-editor.org/info/rfc7201>. [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, DOI 10.17487/RFC7667, November 2015, <https://www.rfc-editor.org/info/rfc7667>. [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017, <https://www.rfc-editor.org/info/rfc8174>. [RFC8285] Singer, D., Desineni, H., and R. Even, Ed., "A General Mechanism for RTP Header Extensions", RFC 8285, DOI 10.17487/RFC8285, October 2017, <https://www.rfc-editor.org/info/rfc8285>. [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, <https://www.rfc-editor.org/info/rfc3711>. [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, DOI 10.17487/RFC5506, April 2009, <https://www.rfc-editor.org/info/rfc5506>. [RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax, "Unicast-Based Rapid Acquisition of Multicast RTP Sessions", RFC 6285, DOI 10.17487/RFC6285, June 2011, <https://www.rfc-editor.org/info/rfc6285>. [RFC6828] Xia, J., "Content Splicing for RTP Sessions", RFC 6828, DOI 10.17487/RFC6828, January 2013, <https://www.rfc-editor.org/info/rfc6828>.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure Real-time Transport Protocol (SRTP)", RFC 6904, DOI 10.17487/RFC6904, April 2013, <https://www.rfc-editor.org/info/rfc6904>. [RFC8126] Cotton, M., Leiba, B., and T. Narten, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 8126, DOI 10.17487/RFC8126, June 2017, <https://www.rfc-editor.org/info/rfc8126>. [SCTE35] Society of Cable Telecommunications Engineers (SCTE), "Digital Program Insertion Cueing Message for Cable", 2016, <http://www.scte.org/SCTEDocs/Standards/ SCTE%2035%202016.pdf>.