| Network Working Group | S. Proust |
| Internet-Draft | Orange |
| Intended status: Informational | E. Berger |
| Expires: April 2, 2015 | Cisco |
| B. Feiten | |
| Deutsche Telekom | |
| B. Burman | |
| Ericsson | |
| K. Bogineni | |
| Verizon Wireless | |
| M. Lei | |
| Huawei | |
| E. Marocco | |
| Telecom Italia | |
| September 29, 2014 |
Additional WebRTC audio codecs for interoperability with legacy networks.
draft-ietf-rtcweb-audio-codecs-for-interop-00
To ensure a baseline level of interoperability between WebRTC clients, [I-D.ietf-rtcweb-audio] requires a minimum set of codecs. However, to maximize the possibility to establish the session without the need for audio transcoding, it is also recommended to include in the offer other suitable audio codecs that are available to the browser.
This document provides some guidelines on the suitable codecs to be considered for WebRTC clients to address the most relevant interoperability use cases.
This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at http://datatracker.ietf.org/drafts/current/.
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."
This Internet-Draft will expire on April 2, 2015.
Copyright (c) 2014 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 (http://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.
As indicated in [I-D.ietf-rtcweb-overview], it has been anticipated that WebRTC will not remain an isolated island and that some WebRTC endpoints will need to communicate with devices used in other existing networks with the help of a gateway. Therefore, in order to maximize the possibility to establish the session without the need for audio transcoding, it is recommended in [I-D.ietf-rtcweb-audio] to include in the offer other suitable audio codecs that are available to the browser. This document provides some guidelines on the suitable codecs to be considered for WebRTC clients to address the most relevant interoperability use cases.
The codecs considered in this document are recommended to be supported and included in the Offer only for WebRTC clients for which interoperability with other non WebRTC end points and non WebRTC based services is relevant as described in sections 5.1.2, 5.2.2 and 5.3.2. Other use cases may justify offering other additional codecs to avoid transcodings. It is the intent of this document to inventory and document any other additional interoperability use cases and codecs if needed.
In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as described in RFC 2119 [RFC2119].
Legacy networks: In this draft, legacy networks encompass the conversational networks that are already deployed like the PSTN, the PLMN, the IMS, H.323 networks.
The mandatory implementation of OPUS [RFC6716] in WebRTC clients can guarantee the codec interoperability (without transcoding) at the state of the art voice quality (better than narrow band "PSTN" quality) only between WebRTC clients. The WebRTC technology is however expected to have more extended usage to communicate with other types of clients. It can be used for instance as an access technology to 3GPP IMS services or to interoperate with fixed or mobile VoIP legacy HD voice service. Consequently, a significant number of calls are likely to occur between terminals supporting WebRTC clients and other terminals like mobile handsets, fixed VoIP terminals, DECT terminals that do not support WebRTC clients nor implement OPUS. As a consequence, these calls are likely to be either of low narrow band PSTN quality using G.711 at both ends or affected by transcoding operations. The drawbacks of such transcoding operations are recalled below:
The following codecs are considered as relevant suitable codecs with respect to the general purpose described in section 4. This list reflects the current status of WebRTC foreseen use cases. It is not limitative and opened to further inclusion of other codecs for which relevant use cases can be identified.
The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP defined speech codec that is mandatory to implement in any 3GPP terminal that supports wideband speech communication. It is being used in circuit switched mobile telephony services and new multimedia telephony services over IP/IMS and 4G/VoLTE, specified by GSMA as voice IMS profile for VoLTE in [IR.92]. More detailed information on AMR-WB can be found in [IR.36]. [IR.36] includes references for all 3GPP AMR-WB related specifications including detailed codec description and Source code.
The market of voice personal communication is driven by mobile terminals. AMR-WB is now implemented in more than 200 devices models and 85 HD mobile networks in 60 countries with a customer base of more than 100 million. A high number of calls are consequently likely to occur between WebRTC clients and mobile 3GPP terminals. The use of AMR-WB by WebRTC clients would consequently allow transcoding free interoperation with all mobile 3GPP wideband terminal. Besides, WebRTC clients running on mobile terminals (smartphones) may reuse the AMR-WB codec already implemented on these devices.
Guidelines for implementing and using AMR-WB and ensuring interoperability with 3GPP mobile services can be found in [TS26.114]. In order to ensure interoperability with 4G/VoLTE as specified by GSMA, the more specific IMS profile for voice derived from [TS26.114] should be considered in [IR.92].
Adaptive Multi-Rate (AMR) is a 3GPP defined speech codec that is mandatory to implement in any 3GPP terminal that supports voice communication, i.e. several hundred millions of terminals. This include both mobile phone calls using GSM and 3G cellular systems as well as multimedia telephony services over IP/IMS and 4G/VoLTE, such as GSMA voice IMS profile for VoLTE in [IR.92]. In addition to impacts listed above, support of AMR can avoid degrading the high efficiency over mobile radio access.
A user of a WebRTC endpoint on a device integrating an AMR module wants to communicate with another user that can only be reached on a mobile device that only supports AMR. Although more and more terminal devices are now "HD voice" and support AMR-WB; there is still a high number of legacy terminals supporting only AMR (terminals with no wideband / HD Voice capabilities) are still used. The use of AMR by WebRTC client would consequently allow transcoding free interoperation with all mobile 3GPP terminals. Besides, WebRTC client running on mobile terminals (smartphones) may reuse the AMR codec already implemented on these devices.
Guidelines for implementing and using AMR with purpose to ensure interoperability with 3GPP mobile services can be found in [TS26.114]. In order to ensure interoperability with 4G/VoLTE as specified by GSMA, the more specific IMS profile for voice derived from [TS26.114] should be considered in [IR.92].
G.722 is an ITU-T defined wideband speech codec. [G.722] was approved by ITU-T in 1988. It is a royalty free codec that is common in a wide range of terminals and end-points supporting wideband speech and requiring low complexity. The complexity of G.722 is estimated to 10 MIPS [EN300175-8] which is 2.5 to 3 times lower than AMR-WB. Especially, G.722 has been chosen by ETSI DECT as the mandatory wideband codec for New Generation DECT with purpose to greatly increase the voice quality by extending the bandwidth from narrow band to wideband. G.722 is the wideband codec required for CAT-iq DECT certified terminal and the V2.0 of CAT-iq specifications have been approved by GSMA as minimum requirements for HD voice logo usage on "fixed" devices; i.e., broadband connections using the G.722 codec.
G.722 is the wideband codec required for DECT CAT-iq terminals. The market for DECT cordeless phones including DECT chipset is more than 150 Millions per year and CAT-IQ is a registered trade make in 47 countries worldwide. G.722 has also been specified by ETSI in [TS181005] as mandatory wideband codec for IMS multimedia telephony communication service and supplementary services using fixed broadband access. The support of G.722 would consequently allow transcoding free IP interoperation between WebRTC client and fixed VoIP terminals including DECT / CAT-IQ terminals supporting G.722. Besides, WebRTC client running on fixed terminals implementing G.722 may reuse the G.722 codec already implemented on these devices.
Guidelines for implementing and using G.722 with purpose to ensure interoperability with Multimedia Telephony services overs IMS can be found in section 7 of [TS26.114]. Additional information of G.722 implementation in DECT can be found in [EN300175-8] and full codec description and C source code in [G.722].
tbd
tbd
tbd
None.
Thanks to Milan Patel for his review.
| [RFC2119] | Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. |