Understanding RTMFP Handshake

(Disclaimer: the protocol description here is not an official specification of RTMFP but just the protocol understanding based on the OpenRTMFP's Cumulus project as well as the IETF presentation slides.)


What is RTMFP? RTMFP (Real-time media flow protocol) allows UDP-based low-latency end-to-end media path between two Flash Player instances. Compared to earlier RTMP-based media path which runs over TCP, this new protocol enables actual real-time communication on the web. Although the end-to-end media path is not always possible when certain types of NATs and firewalls are present, it is possible to do end-to-end media across most residential-type NATs. The end-to-end media path between two Flash Player reduces latency as well as scalability of the service (or server infrastructure) since most heavy media traffic can be sent without going through the hosted server. The UDP transport reduces latency compared to TCP transport even if the media-path is client-server.

Why is understanding RTMFP important? Unlike the earlier RTMP, the new protocol RTMFP is still closed with no open specification available. There have been some attempts at reverse engineering the protocol for interoperability and some official slides explaining the core logic. Understanding the wire-protocol is not important if you are building Flash-based applications that work among each other. However for applications such as Flash-to-SIP gateway or Flash-to-RTSP translator, where you may need to interoperate between RTMFP and SIP/RTP, it is important to understand the wire-protocol in detail. For a Flash-to-SIP gateway incorporating RTMFP from the Flash side in addition to the existing RTMP will enable low-latency UDP media path between the web user and the translator service on the Internet.

The following description is reproduced from a contribution (see rtmfp.py) to my RTMP server project.


An RTMFP session is an end-to-end bi-directional pipe between two UDP transport addresses. A transport address contains an IP address and port number, e.g., "". A session can have one or more flows where a flow is a logical path from one entity to another via zero or more intermediate entities. UDP packets containing encrypted RTMFP data are exchanged in a session. A packet contains one or more messages. A packet is always encrypted using AES with 128-bit keys.

In the protocol description below, all numbers are in network byte order (big-endian). The | operator indicates concatenation of data. The numbers are assumed to be unsigned unless mentioned explicitly.

Scrambled Session ID

The packet format is as follows. Each packet has the first 32 bits of scrambled session-id followed by encrypted part. The scrambled (instead of raw) session-id makes it difficult if not impossible to mangle packets by middle boxes such as NATs and layer-4 packet inspectors. The bit-wise XOR operator is used to scramble the first 32-bit number with subsequent two 32-bit numbers. The XOR operator makes it possible to easily unscramble.
packet := scrambled-session-id | encrypted-part

To scramble a session-id,
scrambled-session-id = a^b^c

where ^ is the bit-wise XOR operator, a is session-id, and b and c are two 32-bit numbers from the first 8 bytes of the encrypted-part.

To unscramble,
session-id = x^y^z

where z is the scrambled-session-id, and b and c are two 32-bit numbers from the first 8 bytes of the encrypted-part.

The session-id determines which session keys are used for encryption and decryption of the encrypted part. There is one exception for the fourth message in the handshake which contains the non-zero session-id but the handshake (symmetric) session keys are used for encryption/decryption. For the handshake messages, a symmetric AES (advanced encryption standard) with 128-bit (16 bytes) key of "Adobe Systems 02" (without quotes) is used. For subsequent in-session messages the established asymmetric session keys are used as described later.


Assuming that the AES keys are known, the encryption and decryption of the encrypted-part is done as follows. For decryption, an initialization vector of all zeros (0's) is used for every decryption operation. For encryption, the raw-part is assumed to be padded as described later, and an initialization vector of all zeros (0's) is used for every encryption operation. The decryption operation does not add additional padding, and the byte-size of the encrypted-part and the raw-part must be same.

The decrypted raw-part format is as follows. It starts with a 16-bit checksum, followed by variable bytes of network-layer data, followed by padding. The network-layer data ignores the padding for convenience.
raw-part := checksum | network-layer-data | padding

The padding is a sequence of zero or more bytes where each byte is \xff. Since it uses 128-bit (16 bytes) key, padding ensures that the size in bytes of the decrypted part is a multiple of 16. Thus, the size of padding is always less than 16 bytes and is calculated as follows:
len(padding) = 16*N - len(network-layer-data) - 1

where N is any positive number to make 0 <= padding-size < 16

For example, if network-layer-data is 84 bytes, then padding is 16*6-84-1=11 bytes. Adding a padding of 11 bytes makes the decrypted raw-part of size 96 which is a multiple of 16 (bytes) hence works with AES with 128-bit key.


The checksum is calculated over the concatenation of network-layer-data and padding. Thus for the encoding direction you should apply the padding followed by checksum calculation and then AES encrypt, and for the decoding direction you should AES decrypt, verify checksum and then remove the (optional) padding if needed. Usually padding removal is not needed because network-layer data decoders will ignore the remaining data anyway.

The 16-bit checksum number is calculated as follows. The concatenation of network-layer-data and padding is treated as a sequence of 16-bit numbers. If the size in bytes is not an even number, i.e., not divisible by 2, then the last 16-bit number used in the checksum calculation has that last byte in the least-significant position (weird!). All the 16-bit numbers are added in to a 32-bit number. The first 16-bit and last 16-bit numbers are again added, and the resulting number's first 16 bits are added to itself. Only the least-significant 16 bit part of the resulting sum is used as the checksum.

Network Layer Data

The network-layer data contains flags, optional timestamp, optional timestamp echo and one or more chunks.
network-layer-data = flags | timestamp | timestamp-echo | chunks ...

The flags value is a single byte containing these information: time-critical forward notification, time-critical reverse notification, whether timestamp is present? whether timestamp echo is present and initiator/responder marker. The initiator/responder marker is useful if the symmetric (handshake) session keys are used for AES, so that it protects against packet loopback to sender.

The bit format of the flags is not clear, but the following applies. For the handshake messages, the flags is \x0b. When the flags' least-significant 4-bits are 1101b then the timestamp-echo is present. The timestamp seems to be always present. For in-session messages, the last 4-bits are either 1101b or 1001b.
flags meaning
0000 1011 setup/handshake
0100 1010 in-session no timestamp-echo (server to Flash Player)
0100 1110 in-session with timestamp-echo (server to Flash Player)
xxxx 1001 in-session no timestamp-echo (Flash Player to server)
xxxx 1101 in-session with timestamp-echo (Flash Player to server)

TODO: looks like bit \x04 indicates whether timestamp-echo is present. Probably \x80 indicates whether timestamp is present. last two bits of 11b indicates handshake, 10b indicates server to client and 01b indicates client to server.

The timestamp is a 16-bit number that represents the time with 4 millisecond clock. The wall clock time can be used for generation of this timestamp value. For example if the current time in seconds is tm = 1319571285.9947701 then timestamp is calculated as follows:
int(time * 1000/4) & 0xffff = 46586
, i.e., assuming 4-millisecond clock, calculate the clock units and use the least significant 16-bits.

The timestamp-echo is just the timestamp value that was received in the incoming request and is being echo'ed back. The timestamp and its echo allows the system to calculate the round-trip-time (RTT) and keep it up-to-date.

Each chunk starts with an 8-bit type, followed by the 16-bit size of payload, followed by the payload of size bytes. Note that \xff is reserved and not used for chunk-type. This is useful in detecting when the network-layer-data has finished and padding has started because padding uses \xff. Alternatively, \x00 can also be used for padding as that is reserved type too!
chunk = type | size | payload

Message Flow

There are three types of session messages: session setup, control and flows. The session setup is part of the four-way handshake whereas control and flows are in-session messages. The session setup contains initiator hello, responder hello, initiator initial keying, responder initial keying, responder hello cookie change and responder redirect. The control messages are ping, ping reply, re-keying initiate, re-keying response, close, close acknowledge, forwarded initiator hello. The flow messages are user data, next user data, buffer probe, user data ack (bitmap), user data ack (ranges) and flow exception report.

A new session starts with an handshake of the session setup. Under normal client-server case, the message flow is as follows:
 initiator (client)                target (server)
|-------initiator hello---------->|
|<------responder hello-----------|

Under peer-to-peer session setup case for NAT traversal, the server acts as a forwarder and forwards the hello to another connected client as follows:
 initiator (client)                forwarder (server)                     target (client)
|-------initiator hello---------->| |
| |---------- forwarded initiator hello-->|
| |<--------- ack ----------------------->|
|<------------responder hello---------------------------------------------|

Alternatively, the server could redirect to another target by supplying an alternative list of target addresses as follows:
 initiator (client)                redirector (server)                     target (client)
|-------initiator hello---------->|
|<------responder redirect--------|
|-------------initiator hello-------------------------------------------->|
|<------------responder hello---------------------------------------------|

Note that the initiator, target, forwarder and redirector are just roles for session setup whereas client and server are specific implementations such as Flash Player and Flash Media Server, respectively. Even a server may initiate an initiator hello to a client in which case the server becomes the initiator and client becomes the target for that session. This mechanism is used for the man-in-middle mode in the Cumulus project.

The initiator hello may be forwarded to another target but the responder hello is sent directly. After that the initiator initial keying and the responder initial keying are exchanged (between the initiator and the responded target directly) to establish the session keys for the session between the initiator and the target. The four-way handshake prevents denial-of-service (DoS) via SYN-flooding and port scanning.

As mentioned before the handshake messages for session-setup use the symmetric AES key "Adobe Systems 02" (without the quotes), whereas in-session messages use the established asymmetric AES keys. Intuitively, the session setup is sent over pre-established AES cryptosystem, and it creates new asymmetric AES cryptosystem for the new session. Note that a session-id is established for the new session during the initial keying process, hence the first three messages (initiator-hello, responder-hello and initiator-initial-keying) use session-id of 0, and the last responder-initial-keying uses the session-id sent by the initiator in the previous message. This is further explained later.

Message Types

The 8-bit type values and their meaning are shown below.
type meaning
\x30 initiator hello
\x70 responder hello
\x38 initiator initial keying
\x78 responder initial keying
\x0f forwarded initiator hello
\x71 forwarded hello response

\x10 normal user data
\x11 next user data
\x0c session failed on client side
\x4c session died
\x01 causes response with \x41, reset keep alive
\x41 reset times keep alive
\x5e negative ack
\x51 some ack
TODO: most likely the bit \x01 indicates whether the transport-address is present or not.

The contents of the various message payloads are described below.

Variable Length Data

The protocol uses variable length data and variable length number. Any variable length data is usually prefixed by its size-in-bytes encoded as a variable length number. A variable length number is an unsigned 28-bit number that is encoded in 1 to 4 bytes depending on its value. To get the bit-representation, first assume the number to be composed of four 7-bit numbers as follows
number = 0000dddd dddccccc ccbbbbbb baaaaaaa (in binary)
where A=aaaaaaa, B=bbbbbbb, C=ccccccc, D=ddddddd are the four 7-bit numbers

The variable length number representation is as follows:
0aaaaaaa (1 byte)  if B = C = D = 0
0bbbbbbb 0aaaaaaa (2 bytes) if C = D = 0 and B != 0
0ccccccc 0bbbbbbb 0aaaaaaa (3 bytes) if D = 0 and C != 0
0ddddddd 0ccccccc 0bbbbbbb 0aaaaaaa (4 bytes) if D != 0

Thus a 28-bit number is represented as 1 to 4 bytes of variable length number. This mechanism saves bandwidth since most numbers are small and can fit in 1 or 2 bytes, but still allows values up to 2^28-1 in some cases.


The initiator-hello payload contains an endpoint discriminator (EPD) and a tag. The payload format is as follows:
initiator-hello payload = first | epd | tag

The first (8-bit) is unknown. The next epd is a variable length data that contains an epd-type (8-bit) and epd-value (remaining). Note that any variable length data is prefixed by its length as a variable length number. The epd is typically less than 127 bytes, so only 8-bit length is enough. The tag is a fixed 128-bit (16 bytes) randomly generated data. The fixed sized tag does not encode its length.
epd = epd-type | epd-value

The epd-type is \x0a for client-server and \x0f for peer-to-peer session. If epd-type is peer-to-peer, then the epd-value is peer-id whereas if epd-type is client-server the epd-value is the RTMFP URL that the client uses to connect to. The initiator sets the epd-value such that the responder can tell whether the initiator-hello is for them but an eavesdropper cannot deduce the identity from that epd. This is done, for example, using an one-way hash function of the identity.

The tag is chosen randomly by the initiator, so that it can match the response against the pending session setup. Once the setup is complete the tag can be forgotten.

When the target receives the initiator-hello, it checks whether the epd is for this endpoint. If it is for "another" endpoint, the initiator-hello is silently discarded to avoid port scanning. If the target is an introducer (server) then it can respond with an responder, or redirect/proxy the message with forwarded-initiator-hello to the actual target. In the general case, the target responds with responder-hello.

The responder-hello payload contains the tag echo, a new cookie and the responder certificate. The payload format is as follows:
responder-hello payload = tag-echo | cookie | responder-certificate

The tag echo is same as the original tag from the initiator-hello but encoded as variable length data with variable length size. Since the tag is 16 bytes, size can fit in 8-bits.

The cookie is a randomly and statelessly generated variable length data that can be used by the responder to only accept the next message if this message was actually received by the initiator. This eliminates the "SYN flood" attacks, e.g., if a server had to store the initial state then an attacker can overload the state memory slots by flooding with bogus initiator-hello and prevent further legitimate initiator-hello messages. The SYN flooding attack is common in TCP servers. The length of the cookie is 64 bytes, but stored as a variable length data.

The responder certificate is also a variable length data containing some opaque data that is understood by the higher level crypto system of the application. In this application, it uses the diffie-hellman (DH) secure key exchange as the crypto system.

Note that multiple EPD might map to the single endpoint, and the endpoint has single certificate. A server that does not care about the man-in-middle attack or does not create secure EPD can generate random certificate to be returned as the responder certificate.
certificate = \x01\x0A\x41\x0E | dh-public-num | \x02\x15\x02\x02\x15\x05\x02\x15\x0E

Here the dh-public-num is a 64-byte random number used for DH secure key exchange.

The initiator does not open another session to the same target identified by the responder certificate. If it detects that it already has an open session with the target it moves the new flow requests to the existing open session and stops opening the new session. The responder has not stored any state so does not need to care. (In our implementation we do store the initial state for simplicity, which may change later). This is one of the reason why the API is flow-based rather than session-based, and session is implicitly handled at the lower layer.

If the initiator wants to continue opening the session, it sends the initiator-initial-keying message. The payload is as follows:
initiator-initial-keying payload = initiator-session-id | cookie-echo
| initiator-certificate | initiator-component | 'X'

Note that the payload is terminated by \x58 (or 'X' character).

The initiator picks a new session-id (32-bit number) to identify this new session, and uses it to demultiplex subsequent received packet. The responder uses this initiator-session-id as the session-id to format the scrambled session-id in the packet sent in this session.

The cookie-echo is the same variable length data that was received in the responder-hello message. This allows the responder to relate this message with the previous responder-hello message. The responder will process this message only if it thinks that the cookie-echo is valid. If the responder thinks that the cookie-echo is valid except that the source address has changed since the cookie was generated it sends a cookie change message to the initiator.

In this DH crypto system, p and g are publicly known. In particular, g is 2, and p is a 1024-bit number. The initiator picks a new random 1024-bit DH private number (x1) and generates 1024-bit DH public number (y1) as follows.
y1 = g ^ x1 % p

The initiator-certificate is understood by the crypto system and contains the initiator's DH public number (y1) in the last 128 bytes.

The initiator-component is understood by the crypto system and contains an initiator-nonce to be used in DH algorithm as described later.

When the target receives this message, it generates a new random 1024-bit DH private number (x2) and generates 1024-bit DH public number (y2) as follows.
y2 = g ^ x2 % p

Now that the target knows the initiator's DH public number (y1) and it generates the 1024-bit DH shared secret as follows.
shared-secret = y1 ^ x2 % p

The target generates a responder-nonce to be sent back to the initiator. The responder-nonce is as follows.
responder-nonce = \x03\x1A\x00\x00\x02\x1E\x00\x81\x02\x0D\x02 | responder's DH public number

The peer-id is the 256-bit SHA256 (hash) of the certificate. At this time the responder knows the peer-id of the initiator from the initiator-certificate.

The target picks a new 32-bit responder's session-id number to demultiplex subsequent packet for this session. At this time the server creates a new session context to identify the new session. It also generates asymmetric AES keys to be used for this session using the shared-secret and the initiator and responder nonces as follows.
decode key = HMAC-SHA256(shared-secret, HMAC-SHA256(responder nonce, initiator nonce))[:16]
encode key = HMAC-SHA256(shared-secret, HMAC-SHA256(initiator nonce, responder nonce))[:16]

The decode key is used by the target to AES decode incoming packet containing this responder's session-id. The encode key is used by the target to AES encode outgoing packet to the initiator's session-id. Only the first 16 bytes (128-bits) are used as the actual AES encode and decode keys.

The target sends the responder-initial-keying message back to the initiator. The payload is as follows.
responder-initial-keying payload = responder session-id | responder's nonce | 'X'

Note that the payload is terminated by \x58 (or 'X' character). Note also that this handshake response is encrypted using the symmetric (handshake) AES key instead of the newly generated asymmetric keys.

When the initiator receives this message it also calculates the AES keys for this session.
encode key = HMAC-SHA256(shared-secret, HMAC-SHA256(responder nonce, initiator nonce))[:16]
decode key = HMAC-SHA256(shared-secret, HMAC-SHA256(initiator nonce, responder nonce))[:16]

As before, only the first 16 bytes (128-bits) are used as the AES keys. The encode key of initiator is same as the decode key of the responder and the decode key of the initiator is same as the encode key of the responder.

When a server acts as a forwarder, it receives an incoming initiator-hello and sends a forwarded-initiator-hello in an existing session to the target. The payload is follows.
forwarded initiator hello payload := first | epd | transport-address | tag

The first 8-bit value is \x22. The epd value is same as that in the initiator-hello -- a variable length data containing epd-type and epd-value. The epd-type is \x0f for a peer-to-peer session. The epd-value is the target peer-id that was received as epd-value in the initiator-hello.

The tag is echoed from the incoming initiator-hello and is a fixed 16 bytes value.

The transport address contains a flag for indicating whether the address is private or public, the binary bits of IP address and optional port number. The transport address is that of the initiator as known to the forwarder.
transport-address := flag | ip-address | port-number

The flag is an 8-bit number with the first most significant bit as 1 if the port-number is present, otherwise 0. The least significant two bits are 10b for public IP address and 01b for private IP address.

The ip-address is either 4-bytes (IPv4) or 16-bytes (IPv6) binary representation of the IP address.

The optional port-number is 16-bit number and is present when the flag indicates so.

The server then sends a forwarded-hello-response message back to the initiator with the transport-address of the target.
forwarded-hello-response = transport-address | transport-address | ...

The payload is basically one or more transport addresses of the intended target, with the public address first.

After this the initiator client directly sends subsequent messages to the responder, and vice-versa.

A normal-user-data message type is used to deal with any user data in the flows. The payload is shown below.
normal-user-data payload := flags | flow-id | seq | forward-seq-offset | options | data

The flags, an 8-bits number, indicate fragmentation, options-present, abandon and/or final. Following table indicates the meaning of the bits from most significant to least significant.
bit   meaning
0x80 options are present if set, otherwise absent
0x20 with beforepart
0x10 with afterpart
0x02 abandon
0x01 final

The flow-id, seq and forward-seq-offset are all variable length numbers. The flow-id is the flow identifier. The seq is the sequence number. The forward-seq-offset is used for partially reliable in-order delivery.

The options are present only when the flags indicate so using the most significant bit as 1. The options are as follows.

TODO: define options

The subsequent data in the fragment may be sent using next-user-data message with the payload as follows:
next-user-data := flags | data

This is just a compact form of the user data when multiple user data messages are sent in the same packet. The flow-id, seq and forward-seq-offset are implicit, i.e., flow-id is same and subsequent next-user-data have incrementing seq and forward-seq-offset. Options are not present. A single packet never contains data from more than one flow to avoid head-of-line blocking and to enable priority inversion in case of problems.


Will update this article in future:
- Fill in the description of the remaining message flows beyond handshake.
- Describe the man-in-middle mode that enables audio/video flowing through the server.

Three Problems in Interoperating with H.264 of Flash Player

H.264 decoding has been part of Flash Player since version 9, but H264 encoding was recently added in version 11. Once Flash Player 11 beta was out I started looking in to integrating video translation in the SIP-RTMP gateway project. For a Flash-to-Flash video conference you do not need to understand the problems related to H.264 in Flash Player because everything is taken care of behind the scenes by Flash Player. Adding H.264 support in the flash-videoio project was relatively straight forward. However if you are building your own translator to interoperate video between Flash Player and some other application you will need to understand these problems.

1) The first problem is that Flash Player doesn't enable H.264 even for decoding if the RTMP connection does not use the new-style "secure" handshake. In the older version handshaking with bytes containing zeros worked, but not when using H.264. Eventually I found about this on reading some open-source-flash (osflash) forum post and incorporated it in my gateway.

2) The H.264 encoder generates some sequence headers (called SPS and PPS) which are essential in decoding the rest of the video data packets. The same is true with AAC audio codec. In particular in live H.264 publish mode, Flash Player generates periodic SPS/PPS packets so the other Flash Player (or SIP phone) can join the call later and still be able to start decoding the stream. However, some existing SIP video phones generate the sequence packets only once at the beginning. The SIP-RTMP gateway needed to be modified to cache the sequence packets received from non-Flash Player client and re-send them with correct timestamp to the Flash Player client that joined the stream late.

3) Looks like Flash Player 11.0 changed something related to buffering of live stream, which causes problems if the SIP side generates multiple slice NALU (primitive data units in H.264) per frame. The Flash Player itself generates one NALU per frame, however some existing SIP video phones (e.g., Bria 3) generate old-style multiple slice per frame and one NALU per slice and cannot be decoded and displayed in Flash Player 11 in live mode. You can read more about the problem. This is not a problem for buffered playback though. (update on 12/12/2011 -- I can verify that this bug has been fixed in Flash Player and video call works fine now between Bria 3 and Flash Player via my SIP-RTMP gateway)

Ekiga SIP phone uses the new-style RTP mechanism for fragmenting a full H.264 frame instead of using multiple slices in H.264 encoding. This can be easily translated to Flash Player and works with my SIP-RTMP gateway. However, Ekiga has another problem in incorrectly interpreting RTP timestamp of received stream which makes it play the stream much slower.

The Philosophy of Open Source

I recently read a book by Henrik Ingo on "Open Life: The Philosophy of Open Source". I strongly recommend software developers as well as technical managers to read it. Here I present some excerpts that I find very interesting in the book.

"If a buyer is willing to pay a lot for it, then a cheap product can be sold at a high price... It is not stupid to ask a high price, but it is to pay it."

"The law of supply and demand can lead to situations that seem strange when common sense is applied to them. ... The oil is no different to the oil that was on sale at a considerably cheaper price just the day before... When supply goes down, the price goes up - even if all else remains equal."

"Often, the kind of stuff branded a trade secret can also be absurdly insignificant, but the important thing is that they don't tell others about it. Today's companies are at least as interested in the things they don't do as the things they pretend to be doing and producing... There's an ominous sense that much of what we do is done with a logic of mean-spiritedness, whether it is in business or in our everyday lives!"

"In a word, Europe's farming policy is based on mean-spiritedness. The subsidies policy is based on farmers agreeing not to produce more food than their agreed quota (to keep the supply low and prices high)."

"The logic of mean-spiritedness that follows from the law of supply and demand, can also be found in all fields of commerce where there is any co called 'immaterial property', including IT, music, film, and other kinds of entertainment, but the most glaring examples of it occur within the world of computers."

"These three demands -- features, quality, and deadline -- would build certain tension into any project. If, for instance, the schedule is too tight, there may not be enough time to include all the features you want. But if a project manager leans on his team, demanding that all the features are included and the deadline be met, then they are compelled to do a rushed job and, inevitably, quality suffers. ... The Open Source community's no-deadlines principle makes excellent sense, and is probably one of the reasons Open Source programs are so successful... One of the most frequently asked questions at the time was, 'When will the next version of Linux be released?' Linus had a stock answer, which was always, 'When it's done.'"

"Why do people do things? The first reason is survival. The second reason is to have social life. And the third reason is that people do things for fun. In that order... Since we work to have fun, to enjoy it, then why do we drive ourselves into the ground trying to meet artificial deadlines?"

"Usually, the vision and business strategies which guide a company are created in the upper echelons of management, after which it's up to the employees to do whatever the boss requires of them...But the principle of 'do whatever you like' would suggest that the management should quit producing the whole vision and business strategies, and focus instead on making it possible for employees to realize their own vision as best they can. (Unfortunately) For many managers such a concept would seem totally alien."

"The lazier the programmer, the more code he writes. .. Typing is too arduous for him, so he writes the code for a word processing program... Because it's too much effort to print out a letter and take it to the postbox, he writes the code for e-mail... So, laziness is a programmer's prime virtue."

"It's not healthy for one's central motivation to be hatred and fear. And what if one day Linux did manage to bring down Microsoft? Would life then lose its meaning? In order to energize themselves, would the programmers then have to find some new and fearful threat to compete against?"

"Since the beginning of hacking, Open Source hackers have always made programs to suit their own needs. ... As a client, the Federal Republic of Germany accepted this logic, and they aren't likely to have any reason to complain. Not only did they get what they wanted, they got a high-quality solution, they got it cheap, and they got it fast. What could be unfair about that?"

"An interesting situation -- IBM had to keep developing Eclipse; yet, financially, investing in it was a bad idea. The solution, of course, was Open Source."

"A company that has calculated its tender openly is much easier to trust. If I were to receive an honest tender of 1,000,000 from a company that operated with open principles, and the tender from a closed company came in at 999,500, I am likely to laugh at the latter and accept the former."

"I once read somewhere about a study which showed that about 20 percent of ants in an anthill do totally stupid things, such as tear down walls the other ants have just built, or take recently gathered food and stow it where none of them will ever find it, or do other things to sabotage everything the other ants so diligently try to achieve. The theory is that these ants don't do these things out of malice but simply because they're stupid... Critics of Open Source projects claim that their non-existent hierarchy and lack of organization leads to inefficiency... If a number of people do some stupid things, we make a rule to say it mustn't be done. Then we need a rule that says everybody has to read the rules. Before long, we need managers and inspectors to make sure people read and follow the rules and that nobody does anything stupid, even by mistake. Finally, the organization has a large group of people who spend time thinking up and writing rules, and enforcing them. And those not involved in doing, are primarily concerned with not breaking the rules...However, Linux and Wikipedia prove the opposite is true... This is particularly true when you factor in that not all planners (managers) are all that smart. Which means organizations risk having their entire staff doing something really inane, because that's what somebody planned. So, it seems better to have a little overlapping and lack of planning, because at least you have better odds for some of the overlapping activities actually making sense..."

Internet Video Communication: Past, Present and Future

I gave a presentation last month titled Hello 1 2 3, can you hear see me now? highlighting my point of view on the origins of Internet video communication technologies we see today. The full text of the presentation can be found at Internet video communication: past, present and future.

Modern video communication systems have roots in several technologies: transporting video over phone lines, using multicast on Internet2's Mbone, adding video to voice-over-IP (VoIP), and adding interactivity in existing streaming applications. Although the Internet telephony and multimedia communication protocols have matured over the last fifteen years, they are largely being used for interconnectivity among closed networks of telecom services. Recently, the world wide web has evolved as a popular platform for everything we do on the Internet including email, text chat, voice calls, discussions, enterprise applications and multi-party collaboration. Unfortunately, there is a disconnect between the web and traditional Internet telephony protocols as they have ignored the constraints and requirements of each other. Consequently, Adobe's Flash Player is being used as a web browser plugin by many developers for voice and video calls over the web.

Learning from the mistakes of the past and knowing where we stand at present will help us build the Internet video communication systems of the future. I present my point of view on the evolution, challenges and mistakes of the past, and, moving forward, describe the challenges in bridging the gap between web and VoIP. I highlight my contributions at various stages in the journey of Internet audio/video communication protocols.