| Transport Area Working Group | J. Saldana, Ed. |
| Internet-Draft | University of Zaragoza |
| Obsoletes: 4170 (if approved) | February 2012 |
| Intended status: Best Current Practice | |
| Expires: August 02, 2012 |
Tunneling Compressed Multiplexed Traffic Flows (TCMTF)
draft-saldana-tsvwg-tcmtf-00
This document describes a method to improve the bandwidth utilization of network paths that carry multiple streams in parallel between two endpoints, as in voice trunking. The method combines standard protocols that provide compression, multiplexing, and tunneling over a network path for the purpose of reducing the bandwidth used when multiple streams are carried over that path.
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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.
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 [RFC2119].
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 [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 [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 (Table 1).
| 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 ([cRTP], [ECRTP], [IPHC], [ROHC]). 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 [FLV]).
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 limited 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.
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 [SCTP]). 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 [TCRTP] defines a way to combine different standard protocols. Three layers are considered, as shown in the figure:
RTP/UDP
|
| ----------------------------
|
ECRTP compressing layer
|
| ----------------------------
|
PPPMUX multiplexing layer
|
| ----------------------------
|
L2TP tunneling layer
|
| ----------------------------
|
IP
In contrast, the new proposal includes other protocols to be compressed in addition to RTP/UDP, since real-time services can also be provided using UDP or TCP.
G.711 or other payload
| ------------------------------
|
G.711.0 or other payload compression payload compression layer
|
| ------------------------------
TCP UDP RTP/UDP
| | |
\ | / ------------------------------
\ | /
Nothing or ROHC or ECRTP or IPHC header compressing layer
|
| ------------------------------
|
PPPMUX or other mux protocols multiplexing layer
|
| ------------------------------
|
GRE or L2TP or other tunneling layer
|
| ------------------------------
IP
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 standard:
[I.711]. This operations can be deployed by network elements like routers/switches, without the endpoints having to signal it using RTSP/SDP/SIP, since G.711 has a fixed RTP payload number.
Finally, another option has been considered: A payload compression layer. When the payload is G.711 this layer can runs G.711.0, a lossless and stateless compression/decompression of the payload
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):
[ending device]---[ending device]
^
|
TCMTF over IP
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.
[ending device]\ /[ending device]
[ending device]---[concentrator]---[concentrator]---[ending device]
[ending device]/ \[ending device]
^ ^ ^
| | |
Native IP TCMTF over IP Native IP
Such a design also allows classical compressing protocols to be used on links with only a few active flows per link.
[ending device]\ /[ending device]
[ending device]---[concentrator]---[concentrator]---[ending device]
[ending device]/ \[ending device]
^ ^ ^
| | |
Compressed TCMTF over IP Compressed
There are different protocols that can be used for compressing real-time flows:
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 [ECRTP]) 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 [ECRTP].
Because the "twice" algorithm described in [ECRTP] 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 Headers checksum described in [ECRTP].
ROHC [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 [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 IP, UDP, and IP 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 header and tunnel header is to multiplex multiple real-time payloads in a single tunneled packet.
To get reasonable bandwidth efficiency using multiplexing within an L2TP 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 L2TP 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 L2TP header. A smaller L2TP header will result in fewer simultaneous sessions being required to produce adequate bandwidth efficiencies.
L2TP tunnels should be used to tunnel the ECRTP payloads end to end. L2TP includes methods for tunneling messages used in PPP session establishment, such as NCP (Network Control Protocol). This allows [IPCP-HC] to negotiate ECRTP compression/decompression parameters.
Other tunneling schemes, such as GRE [GRE] may also be used to implement the tunneling layer of TCMTF.
The packet format for a packet compressed is:
+------------+-----------------------+
| | |
| Compr | |
| Header | Data |
| | |
| | |
+------------+-----------------------+
The packet format of a multiplexed PPP packet as defined by [PPP-MUX] is:
+-------+---+------+-------+-----+ +---+------+-------+-----+
| Mux |P L| | | | |P L| | | |
| PPP |F X|Len1 | PPP | | |F X|LenN | PPP | |
| Prot. |F T| | Prot. |Info1| ~ |F T| | Prot. |InfoN|
| Field | | Field1| | | |FieldN | |
| (1) |1-2 octets| (0-2) | | |1-2 octets| (0-2) | |
+-------+----------+-------+-----+ +----------+-------+-----+
The combined format used for TCMTF with a single payload is all of the above packets concatenated. Here is an example with one payload:
+------+------+-------+----------+-------+--------+----+
| IP |Tunnel| Mux |P L| | | | |
|header|header| PPP |F X|Len1 | PPP | Compr | |
| (20) | | Proto |F T| | Proto | header |Data|
| | | Field | | Field1| | |
| | | (1) |1-2 octets| (0-2) | | |
+------+------+-------+----------+-------+--------+----+
|<------------- IP payload -------------------->|
|<-------- Mux payload --------->|
If the tunnel contains multiplexed traffic, multiple "PPPMux payload"s are transmitted in one IP packet.
Dan Wing
Cisco Systems
771 Alder Drive
San Jose, CA 95035
US
Phone: +44 7889 488 335
Email: dwing@cisco.com
Julian Fernandez-Navajas
University of Zaragoza
Dpt. IEC Ada Byron Building
Zaragoza, 50018
Spain
Phone: +34 976 761 963
Email: navajas@unizar.es
Jose Ruiz-Mas
University of Zaragoza
Dpt. IEC Ada Byron Building
Zaragoza, 50018
Spain
Phone: +34 976762158
Email: jruiz@unizar.es
Muthu Arul Mozhi Perumal
Cisco Systems, Inc.
Cessna Business Park
Sarjapur-Marathahalli Outer Ring Road
Bangalore, Karnataka 560103
India
Phone: +91 9449288768
Email: mperumal@cisco.com
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
Email: Gonzalo.Camarillo@ericsson.com
Michael A. Ramalho Cisco Systems, Inc. 1802 Rue de la Porte Wall Township, NJ 07719-3784 US Phone: +1.732.449.5762 Email: mramalho@cisco.com
This memo includes no request to IANA.
All drafts are required to have a security considerations section. See RFC 3552 [RFC3552] for a guide.
| [RFC3552] | Rescorla, E. and B. Korver, "Guidelines for Writing RFC Text on Security Considerations", BCP 72, RFC 3552, July 2003. |