Internet Draft A. H. Li draft-ietf-avt-evrc-01.txt UCLA February 18, 2000 Editor Expires: July 2001 An RTP Payload Format for EVRC Speech STATUS OF THIS MEMO This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet- Drafts as reference material or to cite them other than as work in progress. The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. ABSTRACT This document describes the RTP payload format for Enhanced Variable Rate Codec (EVRC) Speech. The packet format supports variable interleaving to reduce the effect of packet loss on Speech quality. In additional, the non-interleaving format is also supported. 1 Introduction This document describes how compressed EVRC speech as produced by the EVRC CODEC [1] may be formatted for use as an RTP payload type. A method is provided to interleave the output of the compressor to reduce quality degradation due to lost packets. Furthermore, the sender may choose various interleave settings based on the importance of low end-to-end delay versus greater tolerance for lost packets. The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [3]. 2 Background The Electronic Industries Association (EIA) & Telecommunications Industry Association (TIA) standard IS-127 [1] defines a speech compression algorithm for use in cdma2000 applications. IS-127, or EVRC is the emerging speech codec standard for cdma2000. The EVRC CODEC [1] compresses each 20 milliseconds of 8000 Hz, 16- bit sampled input speech into one of three different size output frames: Rate 1 (171 bits), Rate 1/2 (80 bits), or Rate 1/8 (16 bits). The CODEC chooses the output frame rate based on analysis of the input speech and the current operating mode (either normal or one of several reduced rates). For typical speech patterns, this results in an average output of 4.2 K bits/sec for normal mode and lower for reduced rate modes. 3 RTP/EVRC Packet Format The RTP timestamp is in 1/8000 of a second units. The RTP payload data for the EVRC CODEC has one of the following 2 formats conditional on whether the receiver uses interleaving and/or bundling or sends one codec frame per packet: For the case where interleaving is in use and/or multiple codec data frames are present in a single RTP packet the RTP packet format is as follows: 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RTP Header [2] | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |RR | LLL | NNN | | +-+-+-+-+-+-+-+-+ one or more codec data frames + | .... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ For the case when interleaving is not used and a single codec data frame is present in a single RTP packet the RTP packet format is as follows: 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RTP Header [2] | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | | + one codec data frames +-+-+-+-+-+-+-+-+ | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The RTP header has the expected values as described in [2]. The extension bit is not set and this payload type MUST never have the marker bit set. The codec data frames are aligned on octet boundaries. When interleaving is in use and/or multiple codec data frames are present in a single RTP packet, the timestamp is, as always, that of the oldest data represented in the RTP packet. The other fields have the following meaning: Reserved (RR): 2 bit MUST be set to zero by sender, SHOULD be ignored by receiver. Interleave (LLL): 3 bits MUST have a value between 0 and 5 inclusive. The remaining two values (6 and 7) MUST not be used by senders. If this field is non-zero, interleaving is enabled. All receivers MUST support interleaving. Senders MAY support interleaving. Senders that do not support interleaving MUST set field LLL and NNN to zero. Interleave Index (NNN): 3 bits MUST have a value less than or equal to the value of LLL. Values of NNN greater than the value of LLL are invalid. Interleaving/Bundling indication can be determined at the receiver by detecting the presence of a 1 in the first bit of the RTP packet payload. 3.1 Receiving Invalid Values On receipt of an RTP packet with an invalid value of the LLL or NNN field, the RTP packet MUST be treated as lost by the receiver for the purpose of generating erasure frames as described in section 4. 3.2 CODEC data frame format The output of the EVRC CODEC must be converted into CODEC data frames for inclusion in the RTP payload as follows: a. Octet 0 of the CODEC data frame indicates whether interleaving is present, if rate reduction is desired, and the rate of the codec frame. The format of the octet is indicated below: 0 1 2 3 4 5 6 7 +-+-+-+-+-+-+-+-+ |I|R| frame type| +-+-+-+-+-+-+-+-+ Interleaving Disabled (I): 1 bit This bit indicates whether the interleaving byte is present. This bit MUST be set to 1 if the interleaving byte is missing (i.e., interleaving/bundling is not used), otherwise it MUST be set to 0. Note: if the first bit of the first RTP payload octet is zero this byte is the interleaving byte, otherwise it is octet zero of the EVRC payload. Reduce Rate (R): 1 bit Setting the 'R' bit indicates that this packet is requesting a reduced codec rate for the reverse direction. When the 'R' bit is not set the packet is requesting that the codec resume normal operation. In the case of packet loss the codec should continue to operate in the mode indicated by the last packet received. Frame Type: 6 bits The frame type values are described in the table below and the size of the associated packet is indicated in the table below: Value RATE TOTAL CODEC data frame size (in octets) --------------------------------------------------------- 0 Blank 1 1 1/8 3 3 1/2 11 4 1 23 14 Erasure 1 (SHOULD NOT be transmitted by sender) Receipt of a CODEC data frame with a reserved value in octet 0 MUST be considered invalid data as described in 3.1. All values not listed in the above table MUST be considered reserved. b. The bits as numbered in the standard [1] from highest to lowest are packed into octets. The highest numbered bit (170 for Rate 1, 79 for Rate 1/2 and 15 for Rate 1/8) is placed in the most significant bit (Internet bit 0) of octet 1 of the CODEC data frame, the second highest bit is placed in the second most significant bit of the first octet, the third highest in the third most significant bit of the first octet, and so on. This continues until all of the bits have been placed in the CODEC data frame. The remaining unused bits of the last octet of the CODEC data frame MUST be set to zero (note that this is only applicable to rate 1 frames as the others fit completely into a whole number of octets). Here is a detail of how a Rate 1 frame is converted into a CODEC data frame: Octet 0 of the data frame has value 4 (see table above) indicating the total data frame length (including octet 0) is 23 octets. Bits 169 through 0 from the standard Rate 1 frame are placed as indicated with bits marked with "Z" being set to zero. The Rate 1/8 and 1/2 standard frames are converted similarly but do not require zero padding because they align on octet boundaries. Rate 1 CODEC data frame (bytes 0 - 3) 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | | |1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1| |I|R| 3(Rate 1) |7|6|6|6|6|6|6|6|6|6|6|5|5|5|5|5|5|5|5|5|5|4|4|4| | | | |0|9|8|7|6|5|4|3|2|1|0|9|8|7|6|5|4|3|2|1|0|9|8|7| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Rate 1 CODEC data frame (bytes 20 - 22) 1 1 1 1 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 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |1|1|1|1|1|1|1|1|1| | | | | | | | | | | | | | | | |8|7|6|5|4|3|2|1|0|9|8|7|6|5|4|3|2|1|0|Z|Z|Z|Z|Z| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 3.3 Bundling CODEC data frames As indicated in section 3, more than one CODEC data frame MAY be included in a single RTP packet by a sender. Receivers MUST handle bundles of up to 10 CODEC data frames in a single RTP packet. Furthermore, senders have the following additional restrictions: o MUST not bundle more CODEC data frames in a single RTP packet than will fit in the MTU of the RTP transport protocol. For the purpose of computing the maximum bundling value, all CODEC data frames should be assumed to have the Rate 1 size. o MUST never bundle more than 10 CODEC data frames in a single RTP packet. o Once beginning transmission with a given SSRC and given bundling value, MUST NOT increase the bundling value. If the bundling value needs to be increased, a new SSRC number MUST be used. o MAY decrease the bundling value only between interleave groups (see section 3.4). If the bundling value is decreased, it MUST NOT be increased (even to the original value), although it may be decreased again at a later time. 3.3.1 Determining the number of bundled CODEC data frames Since no count is transmitted as part of the RTP payload and the CODEC data frames have differing lengths, the only way to determine how many CODEC data frames are present in the RTP packet is to examine octet 0 of each CODEC data frame in sequence until the end of the RTP packet is reached. 3.4 Interleaving CODEC data frames Interleaving is meaningful only when more than one CODEC data frame is bundled into a single RTP packet. All receivers MUST support interleaving. Senders MAY support interleaving. Given a time-ordered sequence of output frames from the EVRC CODEC numbered 0..n, a bundling value B, and an interleave value L where n = B * (L+1) - 1, the output frames are placed into RTP packets as follows (the values of the fields LLL and NNN are indicated for each RTP packet): First RTP Packet in Interleave group: LLL=L, NNN=0 Frame 0, Frame L+1, Frame 2(L+1), Frame 3(L+1), ... for a total of B frames Second RTP Packet in Interleave group: LLL=L, NNN=1 Frame 1, Frame 1+L+1, Frame 1+2(L+1), Frame 1+3(L+1), ... for a total of B frames This continues to the last RTP packet in the interleave group: L+1 RTP Packet in Interleave group: LLL=L, NNN=L Frame L, Frame L+L+1, Frame L+2(L+1), Frame L+3(L+1), ... for a total of B frames Senders MUST transmit in timestamp-increasing order. Furthermore, within each interleave group, the RTP packets making up the interleave group MUST be transmitted in value-increasing order of the NNN field. While this does not guarantee reduced end-to-end delay on the receiving end, when packets are delivered in order by the underlying transport, delay will be reduced to the minimum possible. Additionally, senders have the following restrictions: o Once beginning transmission with a given SSRC and given interleave value, MUST NOT increase the interleave value. If the interleave value needs to be increased, a new SSRC number MUST be used. o MAY decrease the interleave value only between interleave groups. If the interleave value is decreased, it MUST NOT be increased (even to the original value), although it may be decreased again at a later time. 3.5 Finding Interleave Group Boundaries Given an RTP packet with sequence number S, interleave value (field LLL) L, and interleave index value (field NNN) N, the interleave group consists of RTP packets with sequence numbers from S-N to S-N+L inclusive. In other words, the Interleave group always consists of L+1 RTP packets with sequential sequence numbers. The bundling value for all RTP packets in an interleave group MUST be the same. The receiver determines the expected bundling value for all RTP packets in an interleave group by the number of CODEC data frames bundled in the first RTP packet of the interleave group received. Note that this may not be the first RTP packet of the interleave group sent if packets are delivered out of order by the underlying transport. On receipt of an RTP packet in an interleave group with other than the expected bundling value, the receiver MAY discard CODEC data frames off the end of the RTP packet or add erasure CODEC data frames to the end of the packet in order to manufacture a substitute packet with the expected bundling value. The receiver MAY instead choose to discard the whole interleave group and play silence. 3.6 Switching from Interleaved/Bundled Mode to Single EVRC CODEC data Frame Per Packet Mode o If both bundling and interleaving have been reduced to a single CODEC data frame per packet then the sender should switch to the non-inter- leaved/non-bundled RTP payload type description. o Once switching transmission from interleaved/bundled packet mode to single CODEC data frame per packet mode, the sender MUST NOT return to interleave/bundling mode without a new SSRC number being used. 3.7 Reconstructing Interleaved Speech Given an RTP sequence number ordered set of RTP packets in an interleave group numbered 0..L, where L is the interleave value and B is the bundling value, and CODEC data frames within each RTP packet that are numbered in order from first to last with the numbers 1..B, the original, time-ordered sequence of output frames from the CODEC may be reconstructed as follows: First L+1 frames: Frame 0 from packet 0 of interleave group Frame 0 from packet 1 of interleave group And so on up to... Frame 0 from packet L of interleave group Second L+1 frames: Frame 1 from packet 0 of interleave group Frame 1 from packet 1 of interleave group And so on up to... Frame 1 from packet L of interleave group And so on up to... Bth L+1 frames: Frame B from packet 0 of interleave group Frame B from packet 1 of interleave group And so on up to... Frame B from packet L of interleave group 3.7.1 Additional Receiver Responsibility Assume that the receiver has begun playing frames from an interleave group. The time has come to play frame x from packet n of the interleave group. Further assume that packet n of the interleave group has not been received. As described in section 4, an erasure frame will be sent to the EVRC CODEC. Now, assume that packet n of the interleave group arrives before frame x+1 of that packet is needed. Receivers SHOULD use frame x+1 of the newly received packet n rather than substituting an erasure frame. In other words, just because packet n wasn't available the first time it was needed to reconstruct the interleaved speech, the receiver SHOULD NOT assume it's not available when it's subsequently needed for interleaved speech reconstruction. 4 Handling lost RTP packets The EVRC CODEC supports the notion of erasure frames. These are frames that for whatever reason are not available. When reconstructing interleaved speech or playing back non-interleaved speech, erasure frames MUST be fed to the EVRC CODEC for all of the missing packets. Receivers MUST use the timestamp clock to determine how many CODEC data frames are missing. Each CODEC data frame advances the timestamp clock EXACTLY 160 counts. Since the bundling value may vary (it can only decrease), the timestamp clock is the only reliable way to calculate exactly how many CODEC data frames are missing when a packet is dropped. Specifically when reconstructing interleaved speech, a missing RTP packet in the interleave group should be treated as containing B erasure CODEC data frames where B is the bundling value for that interleave group. 5 Implementation Issues and Design Rationale 5.1 Interleaving Length The EVRC CODEC interpolates the missing speech content when given an erasure frame. However, the best quality is perceived by the listener when erasure frames are not consecutive. This makes interleaving desirable as it increases speech quality when dropped packets are more likely. On the other hand, interleaving can greatly increase the end-to-end delay. Where an interactive session is desired, the non-interleaved/ non-bundled RTP payload type is recommended. When end-to-end delay is not a concern, a bundling value of at least 4 and an interleave (field LLL) value of 4 or 5 is recommended subject to MTU limitations. The restrictions on senders set forth in sections 3.3 and 3.4 guarantee that after receipt of the first payload packet from the sender, the receiver can allocate a well-known amount of buffer space that will be sufficient for all future reception from the same SSRC value. Less buffer space may be required at some point in the future if the sender decreases the bundling value or interleave, but never more buffer space. This prevents the possibility of the receiver needing to allocate more buffer space (with the possible result that none is available) should the bundling value or interleave value be increased by the sender. Also, were the interleave or bundling value to increase, the receiver could be forced to pause playback while it receives the additional packets necessary for playback at an increased bundling value or increased interleave. 5.2 Outbound Signaling of Interleaving Even though interleaving of the payload data is signaled by the first bit of Octet 0 of codec data frame, it does not prevent using outbound signal to convey this information if desired. One example would be using different payload type to distinguish interleaved payload data from non-interleaved payload data. In such implementations, two payload type can be signaled - one for interleaved packets and the other one for non-interleaved packets. The receiver can process the RTP packets by the payload type, and simply ignore the I bit (Interleaving Disabled) in the payload packets itself. In addition, the receiver can cross check it against the payload type to improve the robustness. 5.3 Byte Alignment of the Codec Data Frames Because the length of the full rate codec output frame is 171 bits, 5 bits of padding are added at the end of the full rate codec data frames to make them byte aligned. One suggested potential change is to shortening the frame type field in the Octec 0 by 3 bits to eliminate the padding bits at the end of the full rate codec data frames. This will not benefit the half and eighth rate frames. Assuming the typical EVRC codec output scenario as follows: Full rate Half rate Eighth rate Percentage 45% 5% 50% Frame Length (byte) 23 11 3 Saving (byte) 1 0 0 It can be estimated that the overall saving this approach will bring is 3.6% ((1 * 45%) / (23 * 45% + 11 * 5% + 3 * 50%)). Note: This number may not be accurate if considering the additional RTP and any other additional headers of the packet. Also, there are the following concerns about this approach: * The bit shifting operation over the full length of the codec data will add the computation complexity of the algorithm; * This payload type is not particularly designed for over-the-air link where efficiency is one of the primiary concerns; * The non-byte-aligned format is orthogonal to the design of the payload format, and there is not enough evidence of application scenarios to justify the diversification of formats. After these considerations, it is decided that the non-byte-aligned format is not taken at this time. 6. The EVRC MIME Type Registration The MIME-name for the EVRC codec is allocated from the IETF tree since EVRC is expected to be a widely used codec for voice-over-IP applications. Media Type Name: audio Media Subtype Name: EVRC Required Parameters: none Optional parameters for RTP mode: ptime: Defined as usual for RTP audio. maxframes: Maximum number of EVRC speech frames in one RTP packet. The receiver may set this parameter in order to limit buffering requirements or delay. Optional parameters for storage mode: none Encoding considerations for RTP mode: see section 4 and section 5 of this document. Encoding considerations for storage mode: The EVRC speech frames are packed into consecutive compound EVRC payloads, see section 4 and section 5. The compound EVRC payloads must be stored in sequential order. This implies that the first octet after payload n must be the first octet of payload (n+1). Furthermore, missing frames and non-received frames during non-speech period must be encapsulated into a compound EVRC payload as blank frames or erasures. Each receiving entity that accepts this MIME type must be able to decode all EVRC coding modes. Security considerations: see section 8 "Security Considerations". Public specification: this document. Additional information for storage mode: Magic number: none File extensions: evc, EVC Macintosh file type code: none Object identifier or OID: none Intended usage: COMMON. It is expected that many VoIP applications (as well as mobile applications) will use this type. 7. Mapping to SDP Parameters Please note that this chapter applies to the RTP mode only. Parameters are mapped to SDP [5] as usual. Example usage in SDP: m = audio 49120 RTP/EVRC 97 a = rtpmap:97 EVRC a = fmtp:97 maxframes = 2 8 Security Considerations RTP packets using the payload format defined in this specification are subject to the security considerations discussed in the RTP specification [2], and any appropriate profile (for example [4]). This implies that confidentiality of the media streams is achieved by encryption. Because the data compression used with this payload format is applied end-to-end, encryption may be performed after compression so there is no conflict between the two operations. A potential denial-of-service threat exists for data encodings using compression techniques that have non-uniform receiver-end computational load. The attacker can inject pathological datagrams into the stream which are complex to decode and cause the receiver to be overloaded. However, this encoding does not exhibit any significant non-uniformity. As with any IP-based protocol, in some circumstances, a receiver may be overloaded simply by the receipt of too many packets, either desired or undesired. Network-layer authentication may be used to discard packets from undesired sources, but the processing cost of the authentication itself may be too high. In a multicast environment, pruning of specific sources may be implemented in future versions of IGMP [6] and in multicast routing protocols to allow a receiver to select which sources are allowed to reach it. 9 Acknowledgements The editor thanks the following authors for contributions to this document: J. D. Villasenor, D.S. Park, J.H. Park, K. Miller, S. C. Greer, D. Leon, N. Leung, K. J. McKay, M. Lioy, T. Hiller, P. J. McCann, M. D. Turner, A. Rajkumar, and Dan Gal. 10 References [1] TIA/EIA/IS-127, "Enhanced Variable Rate Codec, Speech Service Option 3 for Wideband Spread Spectrum Digital Systems", January 1997. [2] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", RFC 1889, January 1996. [3] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [4] Schulzrinne, H., "RTP Profile for Audio and Video Conferences with Minimal Control", RFC 1890, January 1996. [5] M. Handley and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998. [6] Deering, S., "Host Extensions for IP Multicasting", STD 5, RFC 1112, August 1989. 11 Authors' Address Adam H. Li Image Communication Lab Electrical Engineering Department University of California Los Angeles, CA 90095 USA Phone: +1 310 825 5178 EMail: adamli@icsl.ucla.edu John D. Villasenor Image Communication Lab Electrical Engineering Department University of California Los Angeles, CA 90095 USA Phone: +1 310 825 0228 EMail: villa@icsl.ucla.edu Dong-Seek Park Samsung Electronics Suwon, Kyungki 442-742 Korea Phone: +82 31 200 3674 Email: dspark@samsung.com Jeong-Hoon Park Samsung Electronics Suwon, Kyungki 442-742 Korea Phone: +82 31 200 3747 Email: dspark@samsung.com Keith Miller Nokia 6000 Connection Drive Irving, Texas 75039 USA Phone: +1 972 894 4296 Email: keith.miller@nokia.com S. Craig Greer Nokia 6000 Connection Drive Irving, Texas 75039 USA Phone: +1 972 894 4867 Email: craig.greer@nokia.com David Leon Nokia 6000 Connection Drive Irving, Texas 75039 USA Phone: +1 972 374 1860 Email: david.leon@nokia.com Marcello Lioy QUALCOMM, Incorporated 5775 Morehouse Drive San Diego, CA 92121 USA Phone: +1 858 651 8220 Email: mlioy@qualcomm.com Nikolai Leung QUALCOMM, Incorporated 7710 Takoma Ave. Takoma Park, MD 20912 USA Phone: +1 703 346 8351 Email: nleung@qualcomm.com Kyle J. McKay QUALCOMM, Incorporated 5775 Morehouse Drive San Diego, CA 92121-1714 USA Phone: +1 858 587 1121 EMail: kylem@qualcomm.com Tom Hiller Lucent Technologies Room 2F-218 263 Shuman Drive Naperville, IL 60137 USA Phone: +1 630 979 7673 Email: tom.hiller@lucent.com Peter J. McCann Lucent Technologies Room 2Z-305 263 Shuman Drive Naperville, IL 60137 USA Phone: +1 630 713 9359 Email: mccap@lucent.com Michael D. Turner Lucent Technologies Room 2A-203 67 Whippany Rd Whippany, NJ 07981 USA Phone: +1 973 386 3579 Email: mdturner@lucent.com Ajay Rajkumar Lucent Technologies Room 1A-235 67 Whippany Rd Whippany, NJ 07981 USA Phone: +1 973 386 5249 Email: ajayrajkumar@lucent.com Dan Gal Lucent Technologies 67 Whippany Rd Whippany, NJ 07981 USA Phone: +1 973 428 7734 Email: dgal@lucent.com
Datatracker
draft-ietf-avt-evrc-01
Internet-Draft
