Tunneling Compressed Multiplexed Traffic Flows (TCMTF)

This document describes a way to combine existing protocols for compression, multiplexing, and tunneling to save bandwidth for some applications that generate small packets, such as real-time ones or sensor networks.

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.

In the last years we are witnessing the raise of new real-time services that use the Internet for the delivery of interactive multimedia applications. The most common of these services is VoIP, but many others have been developed, and are experiencing a significant growth: videoconferencing, telemedicine, video vigilance, online gaming, etc. The first design of the Internet did not include any mechanism capable of guaranteeing an upper bound for delivery delay, taking into account that the first deployed services were e-mail, file transfer, etc., in which delay is not critical. RTP was first defined in 1996 in order to permit the delivery of real-time contents. Nowadays, although there are a variety of protocols used for signaling real-time flows (SIP , H.323, etc.), RTP has become the standard par excellence for the delivery of real-time content. RTP was designed to work over UDP datagrams. This implies that an IPv4 packet carrying real-time information has to include 40 bytes of headers: 20 for IPv4 header, 8 for UDP, and 12 for RTP. This overhead is significant, taking into account that many real-time services send very small payloads. It becomes even more significant with IPv6 packets, as the basic IPv6 header is twice the size of the IPv4 header (). IPv4 IPv6 IPv4+UDP+RTP: 40 bytes header IPv6+UDP+RTP: 60 bytes header G.711 at 20 ms packetization: 25% header overhead G.711 at 20 ms packetization: 37.5% header overhead G.729 at 20 ms packetization: 200% header overhead G.729 at 20 ms packetization: 300% header overhead In order to mitigate this bad network efficiency, the multiplexing of a number of payloads into a single packet can be considered as a solution. If we have only one flow, the number of samples included in a packet can be increased, but at the cost of adding new packetization delays. However, if a number of flows share the same path between an origin and a destination, a multiplexer can build a bigger packet in which a number of payloads share a common header. A demultiplexer is necessary at the end of the common path, so as to rebuild the packets as they were originally sent, making multiplexing a transparent process for the extremes of the flow. The headers of the original packets can be compressed to save more bandwidth, taking into account that there exist some header compressing standards (, , , ). When different headers are compressed together, tunneling can be used to relieve intermediate routers from the decompression and compression processing.

But there are many real-time applications that do not use RTP. Some of them send UDP packets, e.g. First Person Shooter (FPS) online games, for which latency is very critical. There is also another fact which has to be taken into account: TCP is getting used for media delivery. For many reasons, such as avoiding firewalls, the standard RTP/UDP/IP protocol stack is substituted in many cases by FLV/HTTP/TCP/IP (FLash Video ). There is also another kind of applications which have been reported as real-time using TCP: MMORPGs (Massively Multiplayer Online Role Playing Games), which in some cases have millions of players, thousands of them sharing the same virtual world. They use TCP packets to send the player commands to the server, and also to send to the player's application the characteristics and situation of other gamers' avatars. These games do not have the same interactivity of FPSs, but the quickness and the movements of the player are important, and can decide if they win or lose a fight.

Different scenarios of application can be considered for the tunneling, compressing and multiplexing solution: for example, voice trunking between gateways of different offices of an enterprise. Also, the traffic of the users of an application in a town or a district, which can be multiplexed and sent to the central server. Also Internet cafes are suitable of having many users of the same application (e.g. a game) sharing the same access link. Another interesting scenario are satellite communication links that often manage the bandwidth by limiting the transmission rate, measured in packets per second (pps), to and from the satellite. Applications like VoIP that generate a large number of small packets can easily fill the maximum number of pps slots, limiting the throughput across such links. As an example, a G.729a voice call generates 50 pps at 20 ms packetization time. If the satellite transmission allows 1,500 pps, the number of simultaneous voice calls is limited to 30. This results in poor utilization of the satellite link's bandwidth as well as places a low bound on the number of voice calls that can utilize the link simultaneously. Multiplexing small packets into one packet for transmission would improve the efficiency. Satellite links would also find it useful to multiplex small TCP packets into one packet. This could be especially interesting for compressing TCP ACKs. Other scenario where traffic can be optimized is the transport over a satellite link of M2M/SCADA (Supervisory Control And Data Acquisition) services. M2M terminals are normally equipped with sensing devices which can interface to proximity sensor networks through wireless connections. The terminal can send the collected sensing data using a satellite link connecting to a satellite gateway, which in turn will forward the M2M/SCADA data to the to the processing and control center through Internet. The size of typical M2M application transaction depends on the specific service and it may vary from a minimum of 20 bytes (e.g., tracking and metering in private security) to about 1,000 bytes (e.g., video-surveillance). In this context, TCMTF concepts can be also applied to allow a more efficient use of the available satellite link capacity, matching the requirements demanded by some M2M services. If the case of large sensor deployments is considered, where proximity sensor networks transmit data through different satellite terminals, the use of compression algorithms already available in current satellite systems to reduce the overhead introduced by TCP or UDP and IPv6 protocols is certainly desirable. In addition to this, tunneling and multiplexing functions available from TCMTF allows extending compression functionality throughout the rest the network, to eventually reach the processing and control centers. There is still another interesting use case: desktop or application sharing where the traffic from the server to the client typically consists of the delta of screen updates. Also, the standard for remote desktop sharing emerging for WebRTC in the RTCWEB Working Group is: {something}/SCTP/UDP (Stream Control Transmission Protocol ). In this scenario, SCTP/UDP could be used in other cases: chatting, file sharing and applications related to WebRTC peers. There could be hundreds of clients at a site talking to a server located at a datacenter over a WAN. Compressing, multiplexing and tunneling this traffic could save WAN bandwidth and potentially improve latency.

In conclusion, a standard that multiplexes, compresses and sends packets using a tunnel can be interesting for many enterprises: developers of VoIP systems can include this option in their solutions; or game providers, who can achieve bandwidth savings in their supporting infrastructures. Other fact that has to be taken into account is that the technique not only saves bandwidth but also reduces the number of packets per second, which sometimes can be a bottleneck for a satellite link or even for a network router. If only one stream is tunneled and compressed, then little bandwidth savings will be obtained. In contrast, multiplexing is helpful to amortize the overhead of the tunnel header over many payloads.

The current standard defines a way to reduce bandwidth and pps of RTP traffic, by combining three different standard protocols: Regarding compression, is the selected option. Multiplexing is accomplished using PPP Multiplexing Tunneling is accomplished by using L2TP (Layer 2 Tunneling Protocol ). The three layers are combined as shown in the figure:

In contrast to the current standard , the new proposal allows the compression of other protocols in addition to RTP/UDP, since real-time services are also provided by bare UDP or TCP, as shown in the figure:

Each of the three layers is considered as independent of the other two, i.e. different combinations of protocols can be implemented according to the new proposal: Regarding compression, a number of options can be considered: as different standards are able to compress different headers (, , , ). The one to be used could be selected depending on the traffic to compress and the concrete scenario (packet loss percentage, delay, etc.). It also exists the possibility of having a null header compression, in the case of wanting to avoid traffic compression, taking into account the need of storing a context for every flow and the problems of context desynchronization in certain scenarios. Multiplexing is also accomplished using PPP Multiplexing . Nevertheless, other multiplexing protocols can also be considered. Tunneling is accomplished by using L2TP (Layer 2 Tunneling Protocol ) over IP, GRE (Generic Routing Encapsulation ) over IP, or MPLS (Multiprotocol Label Switching Architecture ). Payload compression schemes could also be used, but they are not the aim of this standard.

This section describes how to combine three protocols: compressing, multiplexing, and tunneling, to save bandwidth for real-time applications.

TCMTF can be implemented in different ways. The most straightforward is to implement it in the devices terminating the real-time streams (these devices can be e.g. voice gateways, or proxies grouping a number of flows):

Another way TCMTF can be implemented is with an external concentration device. This device could be placed at strategic places in the network and could dynamically create and destroy TCMTF sessions without the participation of the endpoints that generate real-time flows.

A number of already compressed flows can also be merged in a tunnel using a concentrator in order to increase the number of flows in a tunnel:

There are different protocols that can be used for compressing real-time flows: IPHC (IP Header Compression ) permits the compression of TCP/IP, UDP/IP and ESP/IP headers (Encapsulating Security Payload ). It has a low implementation complexity. On the other hand, the resynchronization of the context can be slow over long RTT links. It should be used in scenarios presenting very low packet loss percentage. cRTP (compressed RTP ) works the same way as IPHC, but is also able to compress RTP headers. The link layer transport is not specified, but typically PPP is used. For cRTP to compress headers, it must be implemented on each PPP link. A lot of context is required to successfully run cRTP, and memory and processing requirements are high, especially if multiple hops must implement cRTP to save bandwidth on each of the hops. At higher line rates, cRTP's processor consumption becomes prohibitively expensive. cRTP is not suitable over long-delay WAN links commonly used when tunneling, as proposed by this document. To avoid the per-hop expense of cRTP, a simplistic solution is to use cRTP with L2TP to achieve end-to-end cRTP. However, cRTP is only suitable for links with low delay and low loss. Thus, if multiple router hops are involved, cRTP's expectation of low delay and low loss can no longer be met. Further, packets can arrive out of order. ECRTP (Enhanced Compressed RTP ) is an extension of cRTP that provides tolerance to packet loss and packet reordering between compressor and decompressor. Thus, ECRTP should be used instead of cRTP. ROHC (RObust Header Compression ) is able to compress TCP/IP, UDP/IP, ESP/IP and RTP/UDP/IP headers. It is a robust scheme developed for header compression over links with high bit error rate, such as wireless ones. It incorporates mechanisms for quick resynchronization of the context. It includes an improved encoding scheme for compressing the header fields that change dynamically. Its main drawback is that it requires significantly more processing and memory resources than the ones necessary for IPHC or ECRTP. This standard does not determine which of the existing protocols has to be used for the compressing layer. The decision will depend on the scenario, and will mainly be determined by the packet loss probability, RTT, and the availability of memory and processing resources. The standard is also suitable to include other compressing schemes that may be further developed.

When the compressor receives an RTP packet that has an unpredicted change in the RTP header, the compressor should send a COMPRESSED_UDP packet (described in ) to synchronize the ECRTP decompressor state. The COMPRESSED_UDP packet updates the RTP context in the decompressor. To ensure delivery of updates of context variables, COMPRESSED_UDP packets should be delivered using the robust operation described in . Because the "twice" algorithm described in relies on UDP checksums, the IP stack on the RTP transmitter should transmit UDP checksums. If UDP checksums are not used, the ECRTP compressor should use the cRTP Header checksum described in .

ROHC includes a more complex mechanism in order to maintain context synchronization. It has different operation modes and defines compressor states which change depending on link behavior.

Header compressing algorithms require a layer two protocol that allows identifying different protocols. PPP is suited for this, although other multiplexing protocols can also be used for this layer of TCMTF. When header compression is used inside of a tunnel, it will reduce the size of the headers of the IP packet carried in the tunnel. However, the tunnel itself has overhead due to its IP header and the tunnel header (the information necessary to identify the tunneled payload). One way to reduce the overhead of the IP and tunnel headers is to multiplex multiple real-time payloads in a single tunneled packet.

To get reasonable bandwidth efficiency using multiplexing within a tunnel, multiple real-time streams should be active between the source and destination of an L2TP tunnel. The packet size of the real-time streams has to be small in order to permit a good bandwidth saving. If the source and destination of the tunnel are the same as the source and destination of the compressing protocol sessions, then the source and destination must have multiple active real-time streams to get any benefit from multiplexing. Because of this limitation, TCMTF is mostly useful for applications where many real-time sessions run between a pair of endpoints. The number of simultaneous sessions required to reduce the header overhead to the desired level depends on the size of the tunnel header. A smaller tunnel header will result in fewer simultaneous sessions being required to produce adequate bandwidth efficiencies.

Different tunneling schemes can be used for sending end to end the compressed payloads.

L2TP tunnels should be used to tunnel the compressed payloads end to end. L2TP includes methods for tunneling messages used in PPP session establishment, such as NCP (Network Control Protocol). This allows to negotiate ECRTP compression/decompression parameters. Other tunneling schemes, such as GRE may also be used to implement the tunneling layer of TCMTF.

In some scenarios, mainly in operator´s core networks, the use of MPLS is widely deployed as data transport method. The adoption of MPLS as tunneling layer in this proposal intends to natively adapt TCMTF to those transport networks. In the same way that layer 3 tunnels, MPLS paths, identified by MPLS labels, established between Label Edge Routers (LSRs), could be used to transport the compressed payloads within an MPLS network. This way, multiplexing layer must be placed over MPLS layer. Note that, in this case, layer 3 tunnel headers do not have to be used, with the consequent data efficiency improvement.

The packet format for a packet compressed is:

The packet format of a multiplexed PPP packet as defined by is:

The combined format used for TCMTF with a single payload is all of the above packets concatenated. Here is an example with one payload, using L2TP or GRE tunneling:

| |<-------- Mux payload --------->| ]]> If the tunneling technology is MPLS, then the scheme would be:

| |<-------- Mux payload --------->| ]]> If the tunnel contains multiplexed traffic, multiple "PPPMux payload"s are transmitted in one IP packet.

This memo includes no request to IANA.

All drafts are required to have a security considerations section. See RFC 3552 for a guide.