Robust Header Compression Peter J. McCann Internet Draft Tom Hiller Document: draft-mccann-rohc-gehcoarch-00.txt Lucent Technologies Category: Standards Track November, 2000 Requirements and Architecture for Zero-Byte Header Compression Status of this Memo This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026 [Bradner96]. 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. 1. Abstract Efficient transmission of voice over wireless links requires significant engineering effort. Because of the high cost of bandwidth on such links, special techniques for compression of voice data and its transmission over the air have been developed. The compression techniques and the wireless physical layers have been co-designed for maximum spectral efficiency and human perceptual euphony. Voice over IP (VOIP) applications should be able to leverage this engineering effort when used over wireless links. We advocate a "zero-byte header compression" approach to this problem in order to minimize the impact on typical VOIP applications and to maximize the flexibility of the supporting network elements. This document outlines an architectural framework for a wireless VOIP application, including the wireless link layer and its interface to typical IP stack implementations, and discusses the protocol elements that should be standardized between the various components. McCann, Hiller Standards Track - Expires 06/01 1 GEHCOARCH November 2000 2. Introduction Voice over IP (VOIP) promises to change radically the way that telephony services are built and delivered. Integration of voice with the Internet will not just be a change in the way traffic is carried; rather, new types of services will be made possible by the integration of voice with existing Internet applications such as the World Wide Web and e-mail. The key to these new services will be a platform that offers open programmability while offering a transport for VOIP in an integrated, robust, and efficient way. Wireless links offer great challenges to the transport of voice traffic, and significant engineering effort has gone into making them efficient for circuit voice applications. New voice compression algorithms ("codecs"), such as EVRC [TIA-IS127] or AMR [ETSI-AMR] have been developed to minimize the amount of data that must be carried, and special over-the-air channels have been implemented to carry these codecs with a minimum of overhead bits and minimal latency. VOIP flows will be carried inside the Real-Time Protocol (RTP) [Shulzrinne96] on wired links. However, for wireless links, the situation is less clear. The limited bandwidth of wireless links makes it impossible to transmit the entire IP/UDP/RTP header with every packet, as the overhead would be prohibitive. It is possible to compress these headers by transmitting only updates to the fields that change rather than the entire header [Bormann00], but these compression schemes are complex and can never entirely eliminate the overhead due to RTP. Even when the header is compressed down to one byte per frame on average, the impact on spectral capacity is significant [Hiller00]. Also, the variable- sized frames produced by these compression protocols are unsuitable for typical wireless links that support only a limited number of frame sizes. The fundamental reason why these schemes cannot achieve the same efficiency as circuit data is that they discard information that is available at the physical channel layer, including the real- time nature of the traffic, which can assist in reconstructing the RTP header. This document describes an architectural framework that allows such real-time information to be used while not restricting the choice of call control protocol, placement of call feature servers, or mobile station architecture. Cellular wireless technologies will support distinct bearer channels for real-time audio flows versus non-real-time data. Data for TCP, such as web or e-mail traffic, will suffer from the lossy nature of the wireless link unless a link-layer retransmission protocol is used to improve its reliability. Such McCann, Hiller Standards Track - Expires 06/01 2 GEHCOARCH November 2000 a retransmission protocol (called the Radio Link Protocol or RLP in the emerging cellular data networks) does improve reliability but only at the expense of additional buffering and latency. Real-time audio streams cannot tolerate the additional latency, which could be on the order of 2 seconds. For this reason, a separate bearer channel will be used for voice that does no retransmission. This bearer will be very similar to the existing circuit voice channels. The architecture outlined below allows the mobile station to make effective use of this channel for VOIP. The architecture of a mobile station should allow maximum flexibility in its hardware and software choices. Two basic mobile station models have been identified in the wireless data community. A "network model" station is one that is completely integrated, such as a phone plus browser or a palmtop with integrated radio hardware. Such a device usually has a real-time operating system, a DSP chip for processing the audio codec, and an embedded IP stack implementation. In contrast, a "relay model" station is one that is split in two: it consists of a piece of terminal equipment (such as a laptop computer) connected to a piece of radio equipment, usually by a serial connection. The idea is to make use of the mostly stock operating system on the terminal equipment, while "relaying" the data to and from the wireless network via the radio equipment. We take the point of view that VOIP applications must be supported for both kinds of mobile stations. While network model phones will offer a tightly integrated set of services, relay model stations are likely to offer a much more open and programmable environment on the terminal equipment. As these devices evolve we expect the distinction between network and relay models will blur as the wireless device moves closer to the UNIX notion of a "network interface" to a stock operating system, and the operating system evolves to take on more real-time functionality. McCann, Hiller Standards Track - Expires 06/01 3 GEHCOARCH November 2000 3. Reference Architecture Our reference architecture is shown in Figure 1. Other NRT VOIP GEHCO Apps Control------------Control \ \ | \ \ | +-------------IP Protocol | Stack / | / | / Header Strip-------Data Link--+ Peer Regeneration\ Layer System \___ | | \ | | Audio Codec \ +-------------->Physical<----+ Hardware<--->Impl <--+------------------>Channel(s)<--+ Figure 1. Reference architecture for a system implementing zero-byte header compression. The architecture diagram consists of nine components connected to a peer system. Note that we expect zero-byte header compression to be somewhat asymmetric in that it will usually be implemented between a mobile station, where the VOIP and other applications reside, and a peer network entity that is just a data link termination point and a first-hop Internet router. As such, the peer system in the network will likely be missing the audio hardware and codec implementation, and may not participate in the VOIP control. Also, the mobile station may not need to actually perform header stripping and regeneration if its codec implementation is connected directly to the physical channel, which will likely be required to achieve the desired latency guarantees. We name the zero-byte header compression control portion as "GEHCO control" in reference to the Good Enough Header COmpression proposal [Hiller00]. That document serves as a good example for the kinds of negotiation procedures and message formats that could be used for zero-byte header compression, but we believe this architecture to be sufficiently general to accommodate any similar proposal. In the following subsections we discuss each of the architectural elements in turn. The next section will discuss the interfaces between them. McCann, Hiller Standards Track - Expires 06/01 4 GEHCOARCH November 2000 3.1 Non Real-time Components It is important to distinguish between the real-time and non real- time components of Figure 1. This is especially important for a relay model mobile station, as it impacts which elements of stock operating systems can be reused and which must be implemented as new real-time extensions. In this subsection we examine the non real-time components. 3.1.1 VOIP Control The VOIP control component is the implementation of the call signaling protocol, such as SIP [Handley00] or H.323 [ITU-H323]. We make no assumptions on which protocol is used, and we do not require the network-side peer system to contain this element. The mobile station will use one of the VOIP signaling protocols to interact with call feature servers that could be anywhere on the Internet. We assume that this component will open network-layer connections and will have access to the transport endpoint identifiers for the IP/UDP/RTP flow. However we do not require this element to actually process audio data; it will probably be implemented in user-space and would therefore add unpredictable latency to such flows. 3.1.3 IP Protocol Stack Implementation We assume that the mobile station implements an IP protocol stack in conformance with RFC 1122 [Braden89]. Note that such an implementation is usually not capable of supporting hard real-time tasks. 3.1.4 Data Link Layer The data link layer is the interface between the IP protocol stack and the wireless network device. For cdma2000, this will be PPP [TIA-IS835]. For GPRS, this will be LLC [ETSI-LLC], and for UMTS, this will be PDCP [ETSI-PDCP]. For cdma2000, we assume a mostly stock PPP implementation for interaction with the physical channels that support data and perform retransmission. However, because the data link layer is not a hard real-time component, we would not place it on the audio traffic path inside the mobile station. McCann, Hiller Standards Track - Expires 06/01 5 GEHCOARCH November 2000 3.1.5 GEHCO Control The GEHCO control component is responsible for negotiating the use of GEHCO with the peer system and for setting up context information such as the fixed portion of the IP/UDP/RTP header. It will interact with the VOIP control component to acquire these parameters, and will send them across the data link layer to the peer system. It will also interact with the wireless device (possibly through the data link layer) to establish the physical audio channels and will identify the channel to be used when sending context information to the peer system. 3.1.6 Other Non-Real-Time Applications We expect the terminal equipment to be a general-purpose computer and as such will have other applications running. These applications may interact with other components such as the IP protocol stack, but in general will not be hard real-time tasks. These applications must co-exist will all the other components. 3.2 Real-time Components Because we make use of the real-time nature of the physical channel, several components must be implemented as real-time tasks. For a network model phone, this is similar to existing practice: a tightly integrated, real-time operating system on an embedded device schedules the audio sampling and playback to coincide with the physical frame rate of the wireless link. For a relay model terminal, we wish to make use of the audio hardware on the connected terminal equipment. This may require that the components be implemented using special real-time extensions to existing stock operating systems. 3.2.1 Audio Hardware The audio hardware consists of the analog-to-digital (A/D) and digital-to-analog (D/A) converters used for sampling and playing back sound, along with the analog microphones and speakers. In a network model phone this consists of the integrated equipment that is part of the phone. In a relay model terminal it would be the "sound card" or other audio peripheral. To achieve the required hard real-time performance we assume that special software drivers may be required in such relay model terminals. McCann, Hiller Standards Track - Expires 06/01 6 GEHCOARCH November 2000 3.2.2 Codec Implementation The codec implementation converts the sampled audio to and from the special wireless-specific encoding format. For a network model phone, this encoding is carried out on dedicated Digital Signal Processing (DSP) hardware. In a relay model terminal, we assume this is performed on the general purpose CPU of the terminal equipment. 3.2.3 Physical Channel As mentioned before, there will be two physical channels supporting the mobile station: one that runs RLP retransmission, supporting the latency tolerant data applications; and another that resembles a voice circuit. VOIP control signaling will traverse the data-oriented RLP channel, while the voice bearer traffic will traverse the real-time circuit-like channel. Both channels must be available to the upper layers regardless of whether a relay model or network model terminal is used. The voice channel supports real-time traffic and performs no buffering. It will send a frame at precise, periodic intervals, such as 20 milliseconds for cdma2000. The codec implementation must be able to supply frames for the physical channel at exactly this rate. 3.2.4 Header Stripping/Regeneration We expect the codec implementation to be directly connected to the physical channel on the mobile terminal side, and so concrete IP/UDP/RTP headers may not necessarily appear inside the mobile terminal. Therefore, the header stripping/regeneration component is only really necessary on the network side of the zero-byte header compression protocol. This component is drawn next to the data link layer in the diagram, and is responsible for classifying each packet coming down from the IP protocol stack against the fixed IP/UDP/RTP header fields we are attempting to compress. The value of these fields is established by the GEHCO control component and installed into the header stripping component, possibly via the data link layer. Once the header has been stripped this component must schedule the payload for transmission on the physical layer at the appropriate frame interval, according to the sequence number and timestamp received in the header. In the opposite direction, when packets arrive on the network side from the physical channel, this component is responsible for regenerating the proper IP/UDP/RTP header and passing the packet on to the IP protocol stack. It makes use of the physical arrival McCann, Hiller Standards Track - Expires 06/01 7 GEHCOARCH November 2000 time to generate the proper timestamp and sequence number in the RTP header. Because the header stripping/regeneration component is sending and receiving packets from the IP protocol stack, it is at best a soft real-time component. However, it must interact with the physical voice channel, which is a hard real-time component, both to properly record the frame arrival time and to schedule outgoing packets for transmission. If the header stripping/regeneration is implemented in a separate network element from the physical channel, as is likely to be the case in the emerging cellular architectures [TIA-IS835], then this interaction could be accomplished with the proper use of sequence numbers on the interfaces between them so that each physical frame carries the information about precisely when it arrived or when it is to be transmitted. 4. Interfaces In this section we examine the interfaces between the above components. We distinguish between those interfaces that should be implemented as protocols, suitable for standardization in the IETF or elsewhere, and those that should remain Application Programming Interfaces (APIs) that may or may not need to be standardized. 4.1 Protocol Reference Points In terms of new protocols, the interfaces that need to be standardized are listed below. Some of these interfaces are opportunities for IETF protocols, while others should be carried out by other standards-setting organizations. 4.1.1 GEHCO Control to Data Link Layer The GEHCO control component needs to negotiate the use of GEHCO with its peer and convey the static portion of the IP/UDP/RTP header to the peer. This should be done in such a way that the network side is not required to participate in the VOIP control protocol. This means the network side depends on the mobile station to inform it what are the RTP flows that should be classified by the header stripping component as appropriate for sending over the physical voice channel. Rather than create a new network-layer protocol, we advocate using new data link messages between the two systems to convey this information. When the data link layer is PPP, it is appropriate for the IETF to standardize new message types. The GEHCO proposal [Hiller00] is McCann, Hiller Standards Track - Expires 06/01 8 GEHCOARCH November 2000 an attempt at this. For other link layers it may be more appropriate to carry out this work in other standards-setting bodies, although the IETF message formats may get re-used here. 4.1.2 Data Link Layer to Physical Channel Mobile terminals running PPP will typically generate an octet stream that is appropriate for an underlying physical channel running RLP. However, prior to running PPP the mobile terminal must take steps to establish the channel. Also, we require that the terminal be able to dynamically establish and release the voice channels used for real-time audio. For a network model phone this may be supported by APIs within the phone, but for a relay model terminal this signaling needs to be carried out across a serial port. Such signaling is usually the provenance of a modem control protocol ("AT commands") and standardization is probably best carried out in the International Telecommunications Union (ITU). Note that in addition to the usual signaling to establish and release channels, we also need to obtain identifying information for each channel, along with a precise reference for the time at which the first physical voice frame will be transported across the wireless link. This information will be used by the GEHCO control component to communicate the initial timestamp and sequence number offsets to the peer. It must be possible to signal this information during a running PPP session. Also, the precise timing of handoff events in the network must be communicated to the GEHCO control implementation so that it can properly carry out negotiation with the new peer system. Alternatively, if GEHCO state is transferred from a source peer system to a target peer system, the relationship between RTP timestamps and physical layer frames must be preserved. 4.1.2 Physical Channel to Codec or Header Stripping/Regeneration As stated above, the physical channel will interface directly to the codec implementation on the mobile station side and to a header stripping/regeneration process on the network side. For a network model phone, the codec interface may be a proprietary API. However, for a relay model terminal, we must standardize a new way to transport the frames across a serial connection in real-time. This will require that we multiplex the real-time frames with the non-real-time data for PPP. This multiplexing could be carried out with the use of escape characters on the serial interface; again, this work is probably best carried out within the ITU. Any new special characters would need to be properly inserted into the ACCM of the PPP implementation. On the network side, the physical voice channel may be separated from the header stripping/regeneration process by an IP network. McCann, Hiller Standards Track - Expires 06/01 9 GEHCOARCH November 2000 If this is the case then each physical frame must carry a sequence number that indicates the exact frame time that it was received or is to be transmitted over the air. Standardization of such interfaces is best carried out within the 3rd Generation Partnership Projects (3GPPs). 4.2 API Reference Points Other interfaces between the components are best done as Application Programming Interfaces (APIs) and may or may not need to be standardized. In any case we do not advocate the standardization of APIs within the IETF and we discuss these interfaces for illustration purposes only. 4.2.1 VOIP Control to GEHCO Control The VOIP control component is responsible for end-to-end VOIP signaling such as SIP [Handley00] or H.323 [ITU-H323]. We expect these applications to be implemented by many different people and to use standard operating system interfaces. Also, these applications should work the same way when used in wireless or wireline settings, except that the codecs should be tailored for the specific link layer currently in use. When used over wireless links, we expect that applications will want to make use of the optimized real-time path outlined above (audio hardware to codec to physical channel) rather than taking audio data into user space, performing a user space codec transformation, constructing RTP packets, and writing them to a standard UDP socket. Such user space manipulation of audio traffic would introduce unpredictable latency to the flow. To enable the optimized real-time path, the VOIP control protocol should signal to the GEHCO control component that it has completed VOIP signaling and is ready to begin audio bearer flow. This signal might be a system call containing the IP/UDP/RTP parameters that have been negotiated and the codec to be used. This system call would be a one-line addition to existing VOIP client implementations. 4.2.2 GEHCO Control to Real-time Path When the GEHCO control component receives a signal from the VOIP control component that the VOIP signaling has been completed, it must take the following steps: 1) Open the new physical voice bearer channel; McCann, Hiller Standards Track - Expires 06/01 10 GEHCOARCH November 2000 2) Send the peer system information about the flow, including the static header fields and identification of the physical bearer channel; and, finally, 3) Trigger the audio hardware to begin sampling, and the codec implementation to begin encoding/decoding. The first step could be accomplished via an interface to the data link layer, or may be accomplished directly. In any case we need to acquire precise timing information about when the first voice physical frame will be sent and this timing needs to be related to the internal time reference that will be used for RTP timestamps. In the second step, this timing information is used to inform the peer what header fields should be placed on the first physical frame. The third step requires interaction with the real-time components such as the audio hardware and codec implementation, to enable the real-time data to start flowing. 4.2.3 Header Stripping/Regeneration to Data Link Layer The header stripping component must classify all traffic from the IP protocol stack as to whether it is part of the RTP flow that needs to be sent on the voice physical channel. Because it must examine each packet, it will probably be fairly tightly integrated with the data link layer. The header regeneration component produces IP packets from the physical voice frames and sends them up the IP protocol stack. This also may be implemented by passing the packets through the data link layer. 4.2.3 Other Interfaces The mobile terminal potentially will be executing many simultaneous applications and we expect all of the standard interfaces (network sockets, GUI) to be present. Note that ordinary applications may want to use the audio hardware at the same time as a voice call is in progress. This could be disallowed, or a special "audio mixer" process could be introduced between the audio hardware and the codec implementation to allow such simultaneous access. For example, a system beep noise might be mixed into the telephone call in such a way that only the mobile terminal user would hear it. Much ado has been made about the proper reconstruction of the IP Identification field for each RTP packet. We note that RTP payloads are required to stay within the path MTU [Handley99] and should never experience fragmentation. However, in order to avoid any possibility of Identification field collision with other McCann, Hiller Standards Track - Expires 06/01 11 GEHCOARCH November 2000 packets that may be fragmented, a new interface could be implemented between the GEHCO control and the IP protocol stack to "reserve" a range of Identification values for use by the RTP flow. If the header regeneration component always increments the Identification field by one for each reconstructed header, and wraps around to the beginning when the range is about to overflow, then no additional work is necessary to ensure uniqueness of IP Identification fields. 5. Conclusions This draft has presented an architecture for zero-byte header compression and its implications for both a mobile station and the supporting network. On the network side, with this architecture the peer in the network does not need to be aware of the VOIP control between the mobile and a SIP/H323 server that could be anywhere in the network. When the header stripping/regeneration is performed in a network element that is physically separated from the physical channel (e.g. a PDSN from 3GPP2 [TIA-IS835]), the hard real-time requirements on this element can be alleviated through the proper use of sequence numbers on its interface to the radio channel elements. On the mobile side, this draft provides high level requirements for support of zero-byte header compression in the form of protocol interfaces and APIs. Both monolithic network style mobiles as well as relay phone mobiles with laptops are discussed. Proper architecture of the mobile station allows the segregation of hard real-time processing from the non-real-time IP stack and applications. Furthermore, convergence of wireline and wireless applications is a long-standing goal in the wireless industry. This architecture allows mobile end systems to run VOIP based applications developed for wireline access to operate in the wireless environment (although with wireless-specific codecs). The impact on VOIP applications could be as little as one line of code in the VOIP client itself. Finally, the draft has outlined protocol work items suitable for the IETF as well as external standards bodies, including the ITU and 3rd Generation Partnership Projects. Any necessary APIs could be standardized by a collaboration between operating system vendors (open source or otherwise) and third party application developers, driven by wireless service providers. 6. References [Bormann00] Bormann, C. (ed.), "RObust Header Compression (ROHC)," draft-ietf-rohc-rtp-05.txt, October 2000. Work In Progress. McCann, Hiller Standards Track - Expires 06/01 12 GEHCOARCH November 2000 [Braden89] Braden, R. (ed.), "Requirements for Internet Hosts -- Communication Layers," RFC 1122, October 1989. [Bradner96] Bradner, S., "The Internet Standards Process, Revision 3," RFC 2026, October 1996. [ETSI-AMR] European Telecommunications Standards Institute, "Adaptive Multi-Rate (AMR) Speech Transcoding," 3G TS 26.090, February 2000. [ETSI-LLC] European Telecommunications Standards Institute, GSM 04.64. [ETSI-PDCP] European Telecommunications Standards Institute, 3G TS 25.323. [Handley99] Handley, M., and Perkins, C., "Guidelines for Writers of RTP Payload Format Specifications," RFC 2736, December 1999. [Handley00] Handley, Schulzrinne, Schooler, Rosenberg, "SIP: Session Initiation Protocol," draft-ietf-sip- rfc2543bis-01.txt, August 2000. Work In Progress. [Hiller00] Hiller, T., and McCann, P., "Good Enough Header COmpression (GEHCO)," draft-hiller-rohc-gehco- 01.txt, November 2000. Work In Progress. [ITU-H323] International Telecommunications Union, "Packet Based Multimedia Communications Systems," ITU-T Rec. H.323, September 1999. [Shulzrinne96] Schulzrinne, H., Casner, S., Frederick, R., and Jacobson, V., "RTP: A Transport Protocol for Real- Time Applications," RFC 1889, January 1996. [TIA-IS127] Telecommunications Industry Association, "Enhanced Variable Rate Codec, Speech Service 3 for Wideband Spread Spectrum Digital Systems," TIA/EIA/IS-127, February 1997. [TIA-IS835] Telecommunications Industry Association, "Wireless IP Network Standard," TIA/EIA/IS-835, June 2000. 7. Authors' Addresses Peter J. McCann Lucent Technologies Rm 2Z-305 McCann, Hiller Standards Track - Expires 06/01 13 GEHCOARCH November 2000 263 Shuman Blvd Naperville, IL 60566-7050 USA Phone: +1 630 713 9359 FAX: +1 630 713 4982 EMail: mccap@lucent.com Tom Hiller Lucent Technologies Rm 2F-218 263 Shuman Blvd Naperville, IL 60566-7050 USA Phone: +1 630 979 7673 FAX: +1 630 979 7673 EMail: tom.hiller@lucent.com Intellectual Property Statement The IETF takes no position regarding the validity or scope of any intellectual property or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; neither does it represent that it has made any effort to identify any such rights. Information on the IETF's procedures with respect to rights in standards-track and standards-related documentation can be found in BCP-11. 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