Network Working Group JM. Valin Internet-Draft Octasic Inc. Intended status: Standards Track S. Borilin Expires: April 29, 2010 SPIRIT DSP K. Vos Skype C. Montgomery Xiph.Org Foundation R. Chen Broadcom Corporation October 26, 2009 Codec Requirements draft-valin-codec-requirements-02 Status of this Memo This Internet-Draft is submitted to IETF in full conformance with the provisions of BCP 78 and BCP 79. 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. This Internet-Draft will expire on April 29, 2010. Copyright Notice Copyright (c) 2009 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 in effect on the date of publication of this document (http://trustee.ietf.org/license-info). Please review these documents carefully, as they describe your rights Valin, et al. Expires April 29, 2010 [Page 1] Internet-Draft Codec Requirements October 2009 and restrictions with respect to this document. Valin, et al. Expires April 29, 2010 [Page 2] Internet-Draft Codec Requirements October 2009 Abstract This document provides specific requirements for Internet audio codecs. These requirements address quality, sampling rate, bit-rate, and packet loss robustness, as well as other desirable properties. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Applications . . . . . . . . . . . . . . . . . . . . . . . . . 5 2.1. Point to point calls . . . . . . . . . . . . . . . . . . . 5 2.2. Conferencing . . . . . . . . . . . . . . . . . . . . . . . 5 2.3. Telepresence . . . . . . . . . . . . . . . . . . . . . . . 6 2.4. Teleoperation . . . . . . . . . . . . . . . . . . . . . . 6 2.5. In-game voice chat . . . . . . . . . . . . . . . . . . . . 7 2.6. Live distributed music performances / Internet music lessons . . . . . . . . . . . . . . . . . . . . . . . . . 7 2.7. Other applications . . . . . . . . . . . . . . . . . . . . 8 3. Constraints Imposed by the Internet on the Codec . . . . . . . 9 3.1. Security . . . . . . . . . . . . . . . . . . . . . . . . . 10 4. Detailed Basic Requirements . . . . . . . . . . . . . . . . . 11 4.1. Operating space . . . . . . . . . . . . . . . . . . . . . 11 4.2. Quality and bit-rate . . . . . . . . . . . . . . . . . . . 11 4.3. Packet loss robustness . . . . . . . . . . . . . . . . . . 12 4.4. Computational resources . . . . . . . . . . . . . . . . . 12 5. Additional considerations . . . . . . . . . . . . . . . . . . 15 5.1. Low-complexity audio mixing . . . . . . . . . . . . . . . 15 5.2. Encoder side potential for improvement . . . . . . . . . . 15 5.3. Layered bit-stream . . . . . . . . . . . . . . . . . . . . 15 5.4. Partial redundancy . . . . . . . . . . . . . . . . . . . . 16 5.5. Bit error robustness . . . . . . . . . . . . . . . . . . . 16 5.6. Partial redundancy . . . . . . . . . . . . . . . . . . . . 16 5.7. Time stretching and shortening . . . . . . . . . . . . . . 16 5.8. Legacy compatibility . . . . . . . . . . . . . . . . . . . 17 6. Security Considerations . . . . . . . . . . . . . . . . . . . 18 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 19 8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 20 9. Informative References . . . . . . . . . . . . . . . . . . . . 21 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22 Valin, et al. Expires April 29, 2010 [Page 3] Internet-Draft Codec Requirements October 2009 1. Introduction This documents provides requirements for audio codecs designed specifically for use over the Internet. The requirements attempt to address the needs of the most common Internet interactive audio transmission applications and to ensure good quality when operating in conditions that are typical for the Internet. These requirements address the quality, sampling rate, delay, bit-rate, and packet loss robustness. Other desirable codec properties are considered as well. Throughout this document, we will use the following conventions when referring to the sampling rate of a signal: Narrowband: 8 kHz sampling rate Wideband: 16 kHz sampling rate Super-wideband: 32 kHz sampling rate Full-band: 44.1/48 kHz and above Codec bit-rates in bits per second (b/s) will be considered without counting any overhead (IP/UDP/RTP headers, padding, ...). The codec delay is the total algorithmic delay when one adds the codec frame size to the "look-ahead". It is thus the minimum theoretically achievable end-to-end delay of a transmission system that uses the codec. Valin, et al. Expires April 29, 2010 [Page 4] Internet-Draft Codec Requirements October 2009 2. Applications The following applications should be considered for Internet audio codecs, along with their requirements: o Point to point calls o Conferencing o Telepresence o Teleoperation o In-game voice chat o Live distributed music performances / Internet music lessons o Other applications 2.1. Point to point calls Point to point calls are voice over IP (VoIP) calls from two "standard" (fixed or mobile) phones, and implemented in hardware or software. For these applications, a wideband codec is required, along with narrowband support for compatibility with legacy telephony equipment (PSTN). It is expected for the range of useful bit-rates to be 12 - 32 kb/s for wideband speech and 8 - 16 kb/s for narrowband speech. The codec delay must be less than 40 ms, but no more than 25 ms is desirable. Support for encoding music is not required, but it is desirable for the codecs not to make background (on-hold) music excessively unpleasant to hear. Also, the codec should be robust to noise (produce intelligible speech and no annoying artifacts) even at lower bit-rates. 2.2. Conferencing Conferencing applications (which support multi-party calls) have additional requirements on top of the requirements for point-to-point calls. Conferencing systems often have higher-fidelity audio equipment and have greater network bandwidth available -- especially when video transmission is involved. For that reason, support for super-wideband audio becomes important, with useful bit-rates in the 32 - 64 kb/s range. The ability to vary the bit-rate according to the "difficulty" of the audio signal (VBR) is a desirable feature for codecs. This not only saves bandwidth "on average", but it can also help conference servers make more efficient use of the available bandwidth by using more bandwidth for important audio streams and less bandwidth for less important ones (e.g. background noise). Valin, et al. Expires April 29, 2010 [Page 5] Internet-Draft Codec Requirements October 2009 Conferencing end-points often operate in hands-free conditions, which creates acoustic echo problems. For this reason lower delay is important, as it reduces the quality degradation due to any residual echo after acoustic echo cancellation (AEC). For this reason, the codec delay must be less than 30 ms for this application. An optional low-delay mode with less than 10 ms delay is desirable, but not required. Most conferencing systems operate with a bridge that mixes some (or all) of the audio streams and sends them back to all the participants. In that case, it is important that the codec not produce annoying artefacts when two voices are present at the same time. Also, this mixing operation should be as easy as possible to perform. To make it easier to determine which streams have to be mixed (and which are noise/silence), it must be possible to measure (or estimate) the voice activity in a packet without having to fully decode the packet (saving most of the complexity when the packet need not be decoded). Also, the ability to save on the computational complexity when mixing is also desirable, but not required. For example, a transform codec may make it possible to mix the streams in the transform domain, without having to go back to time-domain. Low- complexity up-sampling and down-sampling within the codec is also a desirable feature when mixing streams with different sampling rates. 2.3. Telepresence Most telepresence applications can be considered to be essentially very high-quality video-conferencing environments, so all of the conferencing requirements also apply to telepresence. In addition, telepresence applications require super-wideband and full-band audio capability with useful bit-rates in the 32 - 80 kb/s range. While voice is still the most important signal to be encoded, it must be possible to obtain good quality (even if not transparent) music. Most telepresence applications require more than one audio channel, so support for stereo and multi-channel is important. While this can always be accomplished by encoding multiple single-channel streams, it is preferable to take advantage of the redundancy that exists between channels. 2.4. Teleoperation Teleoperation applications are similar to telepresence, with the exception that they involve remote physical interactions. For example, the user may be controlling a robot while receiving real- time audio feedback from that robot. For these applications, the delay has to be less than 10 ms. The other requirements of telepresence (quality, bit-rate, multi-channel) apply to Valin, et al. Expires April 29, 2010 [Page 6] Internet-Draft Codec Requirements October 2009 teleoperation as well. The only exception is that mixing is not an important issue for teleoperation. 2.5. In-game voice chat An increasing number of computer/console games make use of VoIP to allow players to communicate in real-time. The requirements for gaming are similar to those of conferencing, with the main difference being that narrowband compatibility is not necessary. While for most applications a codec delay up to 30 ms is acceptable, a low-delay (< 10 ms) option is highly desirable, especially for games with rapid interactions. The ability to use VBR (with a maximum allowed bitrate) is also highly desirable because it can significantly reduce the bandwidth requirement for a game server. 2.6. Live distributed music performances / Internet music lessons Live music over the Internet requires extremely low end-to-end delay and is one of the most demanding application for interactive audio transmission. It has been observed that for most scenarios, total end-to-end delays up to 25 ms could be tolerated by musicians, with the absolute limit (where none of the scenarios are possible) being around 50 ms [carot09]. In order to achieve this low delay on the Internet -- either in the same city or a nearby city -- the network propagation time must be taken into account. When also subtracting the delay of the audio buffer, jitter buffer, and acoustic path, that leaves around 2 ms to 10 ms for the total delay of the codec. Considering the speed of light in fiber, every 1 ms reduction in the codec delay increases the range over which synchronization is possible by approximately 200 km. Acoustic echo is expected to be an even more important issue for network music than it is in conferencing, especially considering that the music quality requirements essentially forbid the use of a "nonlinear processor" (NLP) with the AEC. This is another reason why very low delay is essential. Considering that the application is music, the full audio bandwidth (44.1 or 48 kHz sampling rate) must be transmitted with a bit-rate that is sufficient to provide near-transparent to transparent quality. With the current audio coding technology, this corresponds to approximately 64 kb/s to 128 kb/s per channel. As for telepresence, support for two or more channels is often desired, so it would be useful for a codec to be able to take advantage of the redundancy that is often present between audio channels. Valin, et al. Expires April 29, 2010 [Page 7] Internet-Draft Codec Requirements October 2009 2.7. Other applications The above list is by no means a complete list of all applications involving interactive audio transmission on the Internet. However, it is believed that meeting the needs of all these different applications should be sufficient to ensure that most applications not listed will also be met. Valin, et al. Expires April 29, 2010 [Page 8] Internet-Draft Codec Requirements October 2009 3. Constraints Imposed by the Internet on the Codec Packet losses are inevitable on the Internet and dealing with those is one of the most fundamental requirements for an Internet audio codec. While any audio codec can be combined with a good packet loss concealment (PLC) algorithm, the important aspect is what happens on the first packets received _after_ the loss. More specifically, this means that: o it should be possible to interpret the contents of any received packet, irrespective of previous losses as specified in BCP 36 [PAYLOADS]; and o the decoder should re-synchronize as quickly as possible (i.e. the output should quickly converge to the output that would have been obtained if no-loss had occurred). The constraint of being able to decode any packet implies the following considerations for an audio codec: o The size of a compressed frame must be kept smaller than the MTU to avoid fragmentation; o The interpretation of any parameter encoded in the bit-stream must not depend on information contained in other packets. For example, it is not acceptable for a codec to allow signaling a mode change in one packet and assume that subsequent frames will be decoded according to that mode. Although the interpretation of parameters cannot depend on other packets, it is still reasonable to use some amount of prediction across frames, provided that the predictors can resynchronize quickly in case of a lost packet. In this case, it is important to use the best compromise between the gain in coding efficiency and the loss in packet loss robustness due to the use of inter-frame prediction. It is a desirable property for the codecs to allow some real-time control of that trade-off so that it can take advantage of more prediction when the loss rate is small, while being more robust to losses when the loss rate is high. To improve the robustness to packet loss, it would be desirable for the codec to allow an adaptive (data- and network-dependent) amount of side information to help improve audio quality when losses occur. For example, this side information may include the retransmission of certain parameters encoded in the previous frame(s). Another important property of the Internet is that it is mostly a best-effort network, with no guaranteed bandwidth. This means that Valin, et al. Expires April 29, 2010 [Page 9] Internet-Draft Codec Requirements October 2009 the codecs have to be able to vary their output bit-rate dynamically (in real-time), without requiring an out-of-band signaling mechanism, and without causing audible artifacts at the bit-rate change boundaries. Additional desirable features are: o Having the possibility to use smooth bit-rate changes with one byte/frame resolution; o Making it possible for a codec to adapt its bit-rate based on the source signal being encoded (source-controlled VBR) to maximize the quality for a certain _average_ bit-rate. Because the Internet transmits data in bytes, codecs should produce compressed data in integer numbers of bytes. In general, the codec design should take into consideration explicit congestion notification (ECN) and may include features that would improve the quality of an ECN implementation. The IETF has defined a set of application-layer protocols to be used for transmitting real-time transport of multimedia data, including voice. It is thus important for the resulting codecs to be easy to use with these protocols. For example, it must be possible to create an [RTP] payload format that conforms to BCP 36 [PAYLOADS]. If any codec parameters need to be negotiated between end-points, the negotiation should be as easy as possible to carry over SIP/SDP or alternatively over XMPP/Jingle. 3.1. Security Just like for any protocol to be used over the Internet, security is a very important aspect to consider. This goes beyond the obvious considerations of preventing buffer overflows and similar attacks that can lead to denial-of-service or remote code execution. One very important security aspect is to make sure that the decoders have a bounded and reasonable worst-case complexity. This prevents an attacker from causing a DoS by sending packets that are specially crafted to take a very long (or infinite) time to decode. A more subtle aspect is the information leak that can occur when the codec is used over an encrypted channel (e.g. [SRTP]). For example, it was suggested [wright08] that use of source-controlled VBR may reveal some information about a conversation through the size of the compressed packets. This would have to be investigated when standardizing a codec. Valin, et al. Expires April 29, 2010 [Page 10] Internet-Draft Codec Requirements October 2009 4. Detailed Basic Requirements This section summarizes all the constraints imposed by the target applications and by the Internet into a set of actual requirements for codec development. 4.1. Operating space The operating space for the target applications can be divided in terms of delay: most applications require a "medium delay" (20-30 ms), while a few require a "very low delay" (< 10 ms). It makes sense to divide the space based on delay because lowering the delay has a cost in terms of quality vs bit-rate. For medium delay, the resulting codecs must be able to efficiently operate within the following range of bit-rates (per channel): o Narrowband: 8 kb/s to 16 kb/s o Wideband: 12 to 32 kb/s o Super-wideband: 24 to 64 kb/s o Full-band: 32 to 80 kb/s Obviously, a lower-delay codec that can operate in the above range is also acceptable. For very low delay, the resulting codecs will need to operate within the following range of bit-rates (per channel): o Super-wideband: 32 to 80 kb/s o Full-band: 48 to 128 kb/s o (Narrowband and wideband not required) 4.2. Quality and bit-rate The quality of a codec is directly linked to the bit-rate, so these two must be considered jointly. When comparing the bit-rate codecs, the overhead of IP/UDP/RTP headers should not be considered, but any additional bits required in the RTP payload format after the header (e.g. required signalling) should be considered. In terms of quality vs bit-rate, the codecs to be developed must be better than the currently available codecs that satisfy the IPR requirements in the guidelines document, which are: Valin, et al. Expires April 29, 2010 [Page 11] Internet-Draft Codec Requirements October 2009 o For narrowband: Speex (NB), GSM-FR, and iLBC(*) o For wideband: Speex (WB), G.722, G.722.1(*) o For super-wideband: Speex (UWB), G.722.1C(*) The codecs marked with (*) do not meet all the licensing guidelines, but the codecs to be developed should still not perform significantly worse. Quality should be measured for multiple languages, including tonal languages. The case of multiple simultaneous voices (as sometimes happens in conferencing) should be evaluated as well. The comparison with the above codecs assumes that the codecs being compared have similar delay characteristics. The bit-rate required for a certain level of quality may be higher than the referenced codecs in cases where a much lower delay is required. In that case, the increase in bit-rate must be less than the ratio between the delays. It is desirable for the codecs to support source-controlled variable bit-rate (VBR) to take advantage from the fact that different inputs require a different bitrate to achieve the same quality. However, it should still be possible to use the codecs at truely constant bit- rate to ensure that no information leak is possible when using an encrypted channel. 4.3. Packet loss robustness Robustness to packet loss is a very important aspect of any codec to be used on the Internet. Codecs must maintain acceptable quality at loss rates up to 5% and maintain good intelligibility up to 15% loss rate. At any sampling rate, bit-rate, and packet loss rate, the quality must be no less than the quality obtained with the Speex codec or the GSM-FR codec in the same conditions. The actual packet loss "patterns" to be used in testing must be obtained from real packet loss traces collected on the Internet, rather than from loss models. These traces should be representative of the typical environments in which the applications of Section 2 operate. For example, traces related to VoIP calls should consider the loss patterns observed for typical home broadband and corporate connections. 4.4. Computational resources The resulting codecs should be implementable on a wide range of devices, so there should be a fixed-point implementation or at least assurance that a reasonable fixed-point is possible. The computational resources figures listed below are meant to be upper Valin, et al. Expires April 29, 2010 [Page 12] Internet-Draft Codec Requirements October 2009 bounds. Even below these bounds, resources should still be minimized. Any proposed increase in computational resources consumption (e.g. to increase quality) should be carefully evaluated even if the resulting resource consumption is below the upper bound. Having variable complexity would be useful (but not required) in achieving that goal as it would allow trading quality/bit-rate for lower complexity. The computational requirements for real-time encoding and decoding are: o Narrowband should require little CPU resources and be implementable on most DSPs with a 16x16 multiplier (e.g. < 40 MIPS). o Wideband can have a bit more complexity than narrowband, but should still be implementable on a cheap DSP (e.g. < 80 MIPS) o Super-wideband/full-band may require higher complexity, but should be implementable on higher-end DSP (e.g. < 200 MIPS), and if possible also on cheaper DSPs as well. The MIPS values are approximate clock frequencies required for real- time encoding+decoding on a DSP capable of single-cycle MAC operations (16x16 multiplication accumulated into 32 bits). Similar computational requirements apply to floating-point processors. For example Narrowband encoding and decoding should be possible using 40 MHz on a modern x86 CPU (2% of a 2 GHz CPU). For applications that require mixing (e.g. conferencing), it must be possible to estimate the energy of the decoded signal with less than 10% of the complexity figures listed above. In terms of memory use, the codec context/state size required should be no more than 2*R*C bytes in floating-point, where R is the sampling rate and C is the number of channels. For fixed-point, that size should be less than R*C. The scratch space required should also be less than 2*R*C bytes for floating point or less than R*C bytes for fixed-point. The combined codec size and data ROM should be small enough not to cause significant implementation problems. Code size is more difficult to evaluate since it is highly dependent on the architecture, but when implemented on an x86 CPU, the codec should require no more than 100 kB for instructions and constant data. It is the intent to maximize the range of devices on which a codec can be implemented. For this reasons, the reference implementation must not depend on "special hardware features" to be present in order to meet the complexity requirement. However, it might be desirable Valin, et al. Expires April 29, 2010 [Page 13] Internet-Draft Codec Requirements October 2009 to take advantage of such hardware, (e.g., hardware accelerators for operations like FFTs and convolutions). A codec should also minimize the use of saturating arithmetic so as to be implementable on architectures that do not provide hardware saturation (e.g. ARMv4). Valin, et al. Expires April 29, 2010 [Page 14] Internet-Draft Codec Requirements October 2009 5. Additional considerations There are additional features or characteristics that may be desirable under some circumstances, but should not be part of the strict requirements. The benefit of meeting these considerations should be weighted against the associated cost. 5.1. Low-complexity audio mixing In many applications that require a mixing server (e.g. conferencing, games), it is important to minimize the computational cost of the mixing. As much as possible, it should be possible to perform the mixing with fewer computations than it would take to decode all the streams, mix them, and re-encode the result. Properties that reduce the complexity of the mixing process include: o the ability to derive sufficient parameters, such as loudness and/or spectral envelope, for estimating voice activity of a compressed frame without fully decoding that frame; o the ability to mix the streams in an intermediate representation (e.g. transform domain), rather than having to fully decode the signals before the mixing; o the use of bit-stream layers (Section 5.3) by aggregating a small number of active streams at lower quality. For conferencing applications, the total complexity of the decoding, VAD and mixing should be considered when evaluating proposals. 5.2. Encoder side potential for improvement In many codecs, it is possible to improve the quality by improving the encoder without breaking compatibility (i.e. without changing the decoder). Potential for improvement varies from one codec to another. It is generally low for PCM or ADPCM codecs and higher for perceptual transform codecs. All things being equal, being able to improve a codec after the bit-stream is a desirable property. However, this should not be done at the expense of quality in the reference encoder. 5.3. Layered bit-stream A layered codec makes it possible to transmit only a certain subset of the bits and still obtain a valid bit-stream with a quality that is equivalent to the quality that would be obtained from encoding at the corresponding rate. While this is not a necessary feature for most applications, it can be desirable for cases where a "mixing Valin, et al. Expires April 29, 2010 [Page 15] Internet-Draft Codec Requirements October 2009 server" needs to handle a large number of streams with limited computational resources. 5.4. Partial redundancy One possible way of increasing robustness to packet loss is to include partial redundancy within packets. This can be achieved either by including the base layer of the previous frame (for a layered codec) or by transmitting other parameters from the previous frame(s) to assist the PLC algorithm in case of loss. The ability to include partial redundancy for high-loss scenarios is desirable, provided that the feature can be dynamically turned on or off (so that no bandwidth is wasted in case of loss-free transmission). 5.5. Bit error robustness The vast majority of Internet-based applications do not need to be robust to bit errors because packets either arrive unaltered, or do not arrive at all. Considering that, the emphasis should be on packet loss robustness and packet loss concealment. That being said, it is often the case that extra robustness to bit errors can be achieved at no cost at all (i.e. no increase in size, complexity or bit-rate, no decrease in quality or packet loss robustness, ...). In those cases then it is useful to make a change that increases the robustness to bit errors. This can be useful for applications that use UDP Lite transmission (e.g. over a wireless LAN). Robustness to packet loss should *never* be sacrificed to achieve higher bit error robustness. 5.6. Partial redundancy One possible way of increasing robustness to packet loss is to include partial redundancy within packets. This can be achieved either by including the base layer of the previous frame (for a layered codec) or by transmitting other parameters from the previous frame(s) to assist the PLC algorithm in case of loss. The ability to include partial redundancy for high-loss scenarios is desirable, provided that the feature can be dynamically turned on or off (so that no bandwidth is wasted in case of loss-free transmission). 5.7. Time stretching and shortening When adaptive jitter buffers are used it is often necessary to stretch or shorten the audio signal to allow changes in buffering. While this operation can be performed directly on the decoder's output, it is often more computationally efficient to stretch or shorten the signal directly within the decoder. It is desirable for the reference implementation to provide a time stretching/shortening Valin, et al. Expires April 29, 2010 [Page 16] Internet-Draft Codec Requirements October 2009 implementation, although it should not be normative. 5.8. Legacy compatibility In order to create the best possible codec for the Internet, there is no requirement for compatibility with legacy Internet codecs. Valin, et al. Expires April 29, 2010 [Page 17] Internet-Draft Codec Requirements October 2009 6. Security Considerations The codec requirements themselves do not have security considerations. However, codec security issues are discussed in Section 3.1. Valin, et al. Expires April 29, 2010 [Page 18] Internet-Draft Codec Requirements October 2009 7. IANA Considerations This document has no actions for IANA. Valin, et al. Expires April 29, 2010 [Page 19] Internet-Draft Codec Requirements October 2009 8. Acknowledgments We would like to thank all the other people who contributed directly or indirectly to this document, including Jason Fischl, Gregory Maxwell, Alan Duric, Jonathan Christensen, Julian Spittka, and Henry Sinnreich. We also like to thank Cullen Jennings and Gregory Lebovitz for their advice. Valin, et al. Expires April 29, 2010 [Page 20] Internet-Draft Codec Requirements October 2009 9. Informative References [carot09] Carot, A., Werner, C., and T. Fischinger, "Towards a Comprehensive Cognitive Analysis of Delay-Influenced Rhythmical Interaction", 2009. [PAYLOADS] Handley, M. and C. Perkins, "Guidelines for Writers of RTP Payload Format Specifications", RFC 2736, BCP 36. [RTP] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for real-time applications", RFC 3550. [SRTP] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004. [wright08] Wright, C., Ballard, L., Coull, S., Monrose, F., and G. Masson, "Spot me if you can: Uncovering spoken phrases in encrypted VoIP conversations", 2008. Valin, et al. Expires April 29, 2010 [Page 21] Internet-Draft Codec Requirements October 2009 Authors' Addresses Jean-Marc Valin Octasic Inc. 4101, Molson Street Montreal, Quebec Canada Email: jean-marc.valin@octasic.com Slava Borilin SPIRIT DSP Email: borilin@spiritdsp.net Koen Vos Skype Email: koen.vos@skype.net Christopher Montgomery Xiph.Org Foundation Email: xiphmont@xiph.org Raymond (Juin-Hwey) Chen Broadcom Corporation Email: rchen@broadcom.com Valin, et al. Expires April 29, 2010 [Page 22]