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'I-D.webm' Summary: 0 errors (**), 0 flaws (~~), 2 warnings (==), 6 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group C. Bran 3 Internet-Draft Plantronics 4 Intended status: Standards Track C. Jennings 5 Expires: September 13, 2012 Cisco 6 JM. Valin 7 Mozilla 8 March 12, 2012 10 WebRTC Codec and Media Processing Requirements 11 draft-cbran-rtcweb-codec-02 13 Abstract 15 This document outlines the codec and media processing requirements 16 for WebRTC client application and endpoint devices. 18 Status of this Memo 20 This Internet-Draft is submitted in full conformance with the 21 provisions of BCP 78 and BCP 79. This document may not be modified, 22 and derivative works of it may not be created, and it may not be 23 published except as an Internet-Draft. 25 Internet-Drafts are working documents of the Internet Engineering 26 Task Force (IETF). Note that other groups may also distribute 27 working documents as Internet-Drafts. The list of current Internet- 28 Drafts is at http://datatracker.ietf.org/drafts/current/. 30 Internet-Drafts are draft documents valid for a maximum of six months 31 and may be updated, replaced, or obsoleted by other documents at any 32 time. It is inappropriate to use Internet-Drafts as reference 33 material or to cite them other than as "work in progress." 35 This Internet-Draft will expire on September 13, 2012. 37 Copyright Notice 39 Copyright (c) 2012 IETF Trust and the persons identified as the 40 document authors. All rights reserved. 42 This document is subject to BCP 78 and the IETF Trust's Legal 43 Provisions Relating to IETF Documents 44 (http://trustee.ietf.org/license-info) in effect on the date of 45 publication of this document. Please review these documents 46 carefully, as they describe your rights and restrictions with respect 47 to this document. Code Components extracted from this document must 48 include Simplified BSD License text as described in Section 4.e of 49 the Trust Legal Provisions and are provided without warranty as 50 described in the Simplified BSD License. 52 Table of Contents 54 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 55 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 3 56 3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . . 3 57 3.1. Audio Codec Requirements . . . . . . . . . . . . . . . . . 3 58 3.2. Video Codec Requirements . . . . . . . . . . . . . . . . . 3 59 4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . . 4 60 5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . . 5 61 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . . 6 62 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6 63 8. Security Considerations . . . . . . . . . . . . . . . . . . . . 6 64 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 6 65 10. Normative References . . . . . . . . . . . . . . . . . . . . . 6 66 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 7 68 1. Introduction 70 An integral part of the success and adoption of the Web Real Time 71 Communications (WebRTC) will be the voice and video interoperability 72 between WebRTC applications. This specification will outline the 73 media processing and codec requirements for WebRTC client 74 implementations. 76 2. Terminology 78 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 79 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 80 document are to be interpreted as described in RFC 2119 [RFC2119]. 82 3. Codec Requirements 84 This section covers the audio and video codec requirements for WebRTC 85 client applications. To ensure a baseline level of interoperability 86 between WebRTC clients, a minimum set of required codecs are 87 specified below. While this section specifies the codecs that will 88 be mandated for all WebRTC client implementations, it leaves the 89 question of supporting additional codecs to the will of the 90 implementer. 92 3.1. Audio Codec Requirements 94 WebRTC clients are REQUIRED to implement the following audio codecs. 96 o PCMA/PCMU - 1 channel with a rate of 8000 Hz and a ptime of 20 - 97 see section 4.5.14 of [RFC3551] 99 o Telephone Event - [RFC4734] 101 o Opus [draft-ietf-codec-opus] 103 For all cases where the client is able to process audio at a sampling 104 rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before 105 PCMA/PCMU. For Opus, all modes MUST be supported, for all ptime 106 values up to 120 ms. Clients MAY use the offer/answer mechanism to 107 signal a preference for a particular mode or ptime. 109 3.2. Video Codec Requirements 111 The following feature list applies to all required video codecs. 113 Required video codecs: 115 o MUST support at least 10 frames per second (fps) and SHOULD 116 support 30 fps 118 o If VP8 is supported, then it MUST support the bilinear and none 119 reconstruction filters 121 o OPTIONALLY offer support for additional color spaces 123 o MUST support a minimum resolution of 320X240 125 o SHOULD support resolutions of 1280x720, 720x480, 1024x768, 126 800x600, 640x480, 640 x 360 , 320x240 128 4. Audio Level 130 It is desirable to standardize the "on the wire" audio level for 131 speech transmission to avoid users having to manually adjust the 132 playback and to facilitate mixing in conferencing applications. It 133 is also desirable to be consistent with ITU-T recommendations G.169 134 and G.115, which recommend an active audio level of -19 dBm0. 135 However, unlike G.169 and G.115, the audio for WebRTC is not 136 constrained to have a passband specified by G.712 and can in fact be 137 sampled at any sampling rate from 8 kHz to 48 kHz and up. For this 138 reason, the level SHOULD be normalized by only considering 139 frequencies above 300 Hz, regardless of the sampling rate used. The 140 level SHOULD also be adapted to avoid clipping, either by lowering 141 the gain to a level below -19 dBm0, or through the use of a 142 compressor. 144 AUTHORS' NOTE: The idea of using the same level as what the ITU-T 145 recommends is that it should improve inter-operability while at the 146 same time maintaining sufficient dynamic range and reducing the risk 147 of clipping. The main drawbacks are that the resulting level is 148 about 12 dB lower than typical "commercial music" levels and it 149 leaves room for ill-behaved clients to be much louder than a normal 150 client. While using music-type levels is not really an option (it 151 would require using the same compressor-limitors that studios use), 152 it would be possible to have a level slightly higher (e.g. 3 dB) than 153 what is recommended above without causing interoperability problems. 155 Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to 156 a root mean square (RMS) level of 2600. Only active speech should be 157 considered in the RMS calculation. If the client has control over 158 the entire audio capture path, as is typically the case for a regular 159 phone, then it is RECOMMENDED that the gain be adjusted in such a way 160 that active speech have a level of 2600 (-19 dBm0) for an average 161 speaker. If the client does not have control over the entire audio 162 capture, as is typically the case for a software client, then the 163 client SHOULD use automatic gain control (AGC) to dynamically adjust 164 the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop sharing 165 applications, the level SHOULD NOT be automatically adjusted and the 166 client SHOULD allow the user to set the gain manually. 168 The RECOMMENDED filter for normalizing the signal energy is a second- 169 order Butterworth filter with a 300 Hz cutoff frequency. 171 It is common for the audio output on some devices to be "calibrated" 172 for playing back pre-recorded "commercial" music, which is typically 173 around 12 dB louder than the level recommended in this section. 174 Because of this, clients MAY increase the gain before playback. 176 5. Acoustic Echo Cancellation (AEC) 178 It is plausible that the dominant near to mid-term WebRTC usage model 179 will be people using the interactive audio and video capabilities to 180 communicate with each other via web browsers running on a notebook 181 computer that has built-in microphone and speakers. The notebook-as- 182 communication-device paradigm presents challenging echo cancellation 183 problems, the specific remedy of which will not be mandated here. 184 However, while no specific algorithm or standard will be required by 185 WebRTC compatible clients, echo cancellation will improve the user 186 experience and should be implemented by the endpoint device. 188 SHOULD include an AEC and if not, SHOULD ensure that the speaker-to- 189 microphone gain is below unity at all frequencies to avoid 190 instability when none of the client has echo cancellation. For 191 clients that do not control the audio capture and playback devices 192 directly, it is RECOMMENDED to support echo cancellation between 193 devices running at slight different sampling rates, such as when a 194 webcam is used for microphone. 196 The client SHOULD allow either the entire AEC or the non-linear 197 processing (NLP) to be turned off for applications, such as music, 198 that do not behave well with the spectral attenuation methods 199 typically used in NLPs. It SHOULD have the ability to detect the 200 presence of a headset and disable echo cancellation. 202 For some applications where the remote client may not have an echo 203 canceller, the local client MAY include a far-end echo canceller, but 204 if that it the case, it SHOULD be disabled by default. 206 Call control event notification to connected devices such as headsets 207 (what's that exactly?) 209 6. Legacy VoIP Interoperability 211 The codec requirements above will ensure, at a minimum, voice 212 interoperability capabilities between WebRTC client applications and 213 legacy phone systems. 215 7. IANA Considerations 217 This document makes no request of IANA. 219 Note to RFC Editor: this section may be removed on publication as an 220 RFC. 222 8. Security Considerations 224 The codec requirements have no additional security considerations 225 other than those captured in 226 [I-D.ekr-security-considerations-for-rtc-web]. 228 9. Acknowledgements 230 This draft incorporates ideas and text from various other drafts. In 231 particularly we would like to acknowledge, and say thanks for, work 232 we incorporated from Harald Alvestrand. 234 10. Normative References 236 [I-D.ekr-security-considerations-for-rtc-web] 237 Rescorla, E., "Security Considerations for RTC-Web", 238 May 2011. 240 [I-D.webm] 241 Google, Inc., "VP8 Data Format and Decoding Guide", 242 July 2010. 244 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 245 Requirement Levels", BCP 14, RFC 2119, March 1997. 247 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 248 Video Conferences with Minimal Control", STD 65, RFC 3551, 249 July 2003. 251 [RFC4734] Schulzrinne, H. and T. Taylor, "Definition of Events for 252 Modem, Fax, and Text Telephony Signals", RFC 4734, 253 December 2006. 255 Authors' Addresses 257 Cary Bran 258 Plantronics 259 345 Encinial Street 260 Santa Cruz, CA 95060 261 USA 263 Phone: +1 206 661-2398 264 Email: cary.bran@plantronics.com 266 Cullen Jennings 267 Cisco 268 170 West Tasman Drive 269 San Jose, CA 95134 270 USA 272 Phone: +1 408 421-9990 273 Email: fluffy@cisco.com 275 Jean-Marc Valin 276 Mozilla 277 650 Castro Street 278 Mountain View, CA 94041 279 USA 281 Email: jmvalin@jmvalin.ca