Multicasting Strategies: Understanding Reliability

Brad Huntting, University of Colorado
David Mertz, Gnosis Software
April 2003

In the first and second installments we introduced the basic concepts of multicasting and issues of network topology. In this installment Brad and David discuss how the network architecture of multicast can be combined with reliability assurances for the transmitted datastreams.

What Is Reliability?

One of the thorny issues Internet multicast faces is reliability. Reliability, in digital transmission contexts, has three aspects: Assuring the data arrives at its destination:

(a) uncorrupted;
(b) in sequence;
(c) in a timely manner.

However, in the connectionless model of the Internet, packets are delivered on a "best effort" basis, and there is no guarantee about when, or even if, an individual packet will arrive at its destination. How can higher-layer mechanism make up for the lack of reliability at lower network layers?

In reality, it is impossible to make reliability guarantees in all possible network failure scenario; at base, reliability is actually a question of preferred failure modes. Which do you prefer, for example: An intact data stream with no holes, or a time limit on network delay? Do you mind if a few bits are flipped, or would you rather drop a whole packet? Do you want packets delivered in order, or just delivered as fast as possible?

As an example, music should be played at a constant tempo and missing data should produce synchronized audio drop out. This failure mode is the most acceptable degradation of output. Many common audio players, nonetheless, have improperly optimized failure modes--they allow unpleasant failure characteristics, such as skipping ahead when they encounter holes in streaming data.

Transmission Control Protocol And Realtime Transport Protocol

In the unicast world, reliability is usually provided by Transmission Control Protocol (TCP). TCP provides multiple coherently-ordered, error-checked datastreams between two hosts. TCP works by active acknowledgment: each side of the connection notifies the other of how much sequential data it has received. In this way, a lost packet can easily be detected and retransmitted. TCP also detects network congestion, and takes steps to ensure that a socket uses only its share of available bandwidth.

"Classic" multicast multimedia applications such as "vic" and "vat" (multicast video and audio conferencing tools respectively), use the Realtime Transport Protocol (RTP) to organize their data streams. RTP is non-reliable in the sense that it has no mechanism to recover data that is lost in transit. Where TCP uses a 32 bit sequence number (i.e. a byte count) to maintain the order of the packets, RTP uses a 32 bit relative fixed point timestamp which can be used to ensure that data is decoded in the correct order and presented at the correct time. A session-identifier field helps label disparate media streams such as audio and video tracks, so they can be grouped together and synchronized using the timestamps. RTP is typically run over User Datagram Protocol (UDP), which provides a data integrity check.

Environmental Requirements

Many applications require more reliability aspects than just correct ordering and per packet data integrity. Suppose a daily file transfer sends a 500 megabyte file to a million recipients, once a night. To unicast such a file would require around 50 Gb/s of bandwith--any way you cut it, that's expensive. With multicast this transmission could, in theory, be carried on a single 64 Kb/s ISDN. The trick is making sure that the file arrives intact at every receiver.

Adapting TCP's ACK scheme has problems. If every receiver sent ACKs directly to the sender, the return ACK traffic could grow linearly with the number of receivers--this is called an "ACK implosion." If the receivers were to back off, and send ACKs less frequently as the number of listeners increased, then the ACKs would become too spaced-out to be useful. Having receivers send negative acknowledgments (NACKs) when they detected missing data is one solution. But care must be taken to ensure that a packet dropped near the root of the multicast distribution tree does not create a NACK implosion, should all the recipients decide to NACK simultaneously.

Reliable Multicast Transport Working Group Guidelines

To establish a set of standard reliable multicast transport protocols, the Reliable Multicast Transport Working Group (RMT) of the Internet Engineering Task Force[1] created a set of guidelines that developers should address in any proposed protocol. Once implemented, they will, hopefully, be of general use to a wide variety of applications

It was clear from the RMT's outset that different applications would require different aspects of reliability. Unlike the unicast world where TCP is nearly universal, it is unlikely that any single reliable multicast transport protocol will be acceptable for all applications. So the working group has focused on creating protocol modules which can be plugged together to provide different forms of reliability.

The RMT's current focus is on three types of modules:

* Nack Oriented Reliable Multicast protocol (NORM) uses negative acknowledgments for reliability. It avoids the NACK implosion problem by having all recipients listen to NACKs, and/or by using a probabilistic delays to void redundant NACKs. NORM is capable of working over both any source and single source multicast. While NORM can carry so-called non-finite data streams, it will probably be most useful in batch file transfer applications.
* TRee ACKnowledment based protocol (TRACK) uses a tree for controlling feedback and repairs. TRACK has the capability of giving the sender a confirmation receipt for each receiver. This can be important, for example, in financial applications in which the receivers pay for the data--the sender needs confirmation that it can charge the recipients for the data. TRACK may not be a good choice for networks where the return path from receiver to sender is very limited, such as satellite based networks.
* Asynchronous Layered Coding (ALC) uses forward error-correction (FEC) techniques and does not require any feedback. Forward error correction is a technique wherein the sender encodes a message of N symbols (such as bits or packets) as N+P symbols which are transmitted to the receiver. The receiver needs only recover N of the symbols to reconstruct the original message; up to P symbols may be lost or corrupted in transit. In practice, most FEC schemes transmit the N original packets as-is, and create additional parity packets which can be used to recover some of those packets, should they be lost.

Relay Networks And The Their Successor

The currently-limited state of multicast deployment has kept reliable multicast out of the limelight. Fortunately, progress on tools and standardization has continued, and is accelerating.

Currently, application architects needing a mechanism to reliably deliver data to a diverse group of receivers on the Internet have been forced to create distribution trees using unicast links between participating receivers. This technique is called "Application Multicast"; it is used by applications such as Icecast, Gnutella, and Freenet[2], and packages such as Xbone[3]. Application multicast usually works by creating a (virtual) overlay network using unicast Internet links. The main advantage of this approach--and the reason it is used at all--is that the implementation expense is borne by the users. The participating nodes in the overlay network take responsibility for replicating data and constructing the distribution trees.

The performance of the overlay network can be severely degraded when users are behind slow DSL or Cable modem links; this situation is compounded by the asymmetric bandwidth capabilities of such links. Getting the topology of an overlay network correct is no easy task, especially when nodes are allowed to come and go frequently. The overlay network needs not be precisely homomorphic to the underlying Internet topology. But it does need to avoid network bottlenecks, and at the same time avoid excessive fan-out at individual nodes. These are tractable problems, but they are usually simply ignored. The problems can be demonstrated by running a Gnutella client overnight on a consumer-level Internet connection; by morning your Internet connection will be saturated.


1 Information about the Reliable Multicast Transport working group as well as Drafts and RFCs on RMT, can be found at the IETF web site at:

2 Icecast was designed as a free alternative to real audio, info can be found at:

Gnutella is a free peer-to-peer file sharing database, more info can be found by searching for gnutella on Freenet is another, potentially more robust file sharing network, more info can be found on sourceforge at:

3 Xbone, developed at ISI, creates and manages overlay networks or Virtual Private Networks (VPNs) using a variety of tools. More info can be found on their web site at:

Intel Executive Vice President, Leslie Vadasz, in his 2002 Senate Testimony made many good points about the inevitable convergence of Intel PC and network technology with audio and video content distribution. From a technical perspective, research and standardization of multicast protocols and technologies is the essential open matter in furthering such distribution. Vadasz' very interesting statement can be found at:

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