MOPS M.P. Sharabayko
Internet-Draft M.A. Sharabayko
Intended status: Standards Track Haivision Network Video, GmbH
Expires: 10 September 2020 J. Dube
Haivision
JS. Kim
JW. Kim
SK Telecom Co., Ltd.
9 March 2020
The SRT Protocol
draft-sharabayko-mops-srt-00
Abstract
This document specifies Secure Reliable Transport (SRT) protocol.
SRT is a user-level protocol over User Datagram Protocol and provides
reliability and security optimized for low latency live video
streaming, as well as generic bulk data transfer. For this, SRT
introduces control packet extension, improved flow control, enhanced
congestion control and a mechanism for data encryption.
Note to Readers
Source for this draft and an issue tracker can be found at
https://github.com/haivision/srt-rfc (https://github.com/haivision/
srt-rfc).
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
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."
This Internet-Draft will expire on 10 September 2020.
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Copyright Notice
Copyright (c) 2020 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://trustee.ietf.org/
license-info) in effect on the date of publication of this document.
Please review these documents carefully, as they describe your rights
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Motivation . . . . . . . . . . . . . . . . . . . . . . . 3
1.2. Secure Reliable Transport Protocol . . . . . . . . . . . 4
2. Conventions and Definitions . . . . . . . . . . . . . . . . . 5
3. Packet Structure . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Data Packets . . . . . . . . . . . . . . . . . . . . . . 6
3.2. Control Packets . . . . . . . . . . . . . . . . . . . . . 8
3.2.1. Handshake . . . . . . . . . . . . . . . . . . . . . . 9
3.2.2. Keep-Alive . . . . . . . . . . . . . . . . . . . . . 17
3.2.3. ACK (Acknowledgement) . . . . . . . . . . . . . . . . 18
3.2.4. NAK (Loss Report) . . . . . . . . . . . . . . . . . . 20
3.2.5. Shutdown . . . . . . . . . . . . . . . . . . . . . . 21
3.2.6. ACKACK . . . . . . . . . . . . . . . . . . . . . . . 22
4. SRT Data Transmission and Control . . . . . . . . . . . . . . 23
4.1. Stream Multiplexing . . . . . . . . . . . . . . . . . . . 23
4.2. Data Transmission Modes . . . . . . . . . . . . . . . . . 23
4.2.1. Message Mode . . . . . . . . . . . . . . . . . . . . 23
4.2.2. Live Mode . . . . . . . . . . . . . . . . . . . . . . 24
4.2.3. Buffer Mode . . . . . . . . . . . . . . . . . . . . . 24
4.3. Handshake Messages . . . . . . . . . . . . . . . . . . . 24
4.3.1. Caller-Listener Handshake . . . . . . . . . . . . . . 28
4.3.2. Rendezvous Handshake . . . . . . . . . . . . . . . . 30
4.4. SRT Buffer Latency . . . . . . . . . . . . . . . . . . . 36
4.5. Timestamp Based Packet Delivery . . . . . . . . . . . . . 37
4.5.1. Packet Delivery Time . . . . . . . . . . . . . . . . 38
4.6. Too-Late Packet Drop . . . . . . . . . . . . . . . . . . 40
4.7. Drift Management . . . . . . . . . . . . . . . . . . . . 41
4.8. Acknowledgement and Lost Packet Handling . . . . . . . . 42
4.8.1. Packet Acknowledgement (ACKs, ACKACKs) . . . . . . . 43
4.8.2. Packet Retransmission (NAKs) . . . . . . . . . . . . 44
4.9. Bidirectional Transmission Queues . . . . . . . . . . . . 45
4.10. Round Trip Time Estimation . . . . . . . . . . . . . . . 45
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4.11. Congestion Control . . . . . . . . . . . . . . . . . . . 45
5. Encryption . . . . . . . . . . . . . . . . . . . . . . . . . 46
6. Security Considerations . . . . . . . . . . . . . . . . . . . 46
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 46
Contributors . . . . . . . . . . . . . . . . . . . . . . . . . . 46
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 47
References . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
Normative References . . . . . . . . . . . . . . . . . . . . . 47
Informative References . . . . . . . . . . . . . . . . . . . . 47
Appendix A. Packet Sequence List coding . . . . . . . . . . . . 49
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 49
1. Introduction
1.1. Motivation
The demand for live video streaming has been increasing steadily for
many years. With the emergence of cloud technologies, many video
processing pipeline components have transitioned from on-premises
appliances to software running on cloud instances. While real-time
streaming over TCP-based protocols like RTMP[RTMP] is possible at low
bitrates and on a small scale, the exponential growth of the
streaming market has created a need for more powerful solutions.
To improve scalability on the delivery side, content delivery
networks (CDNs) at one point transitioned to segmentation-based
technologies like HLS (HTTP Live Streaming)[RFC8216] and DASH
(Dynamic Adaptive Streaming over HTTP)[ISO23009]. This move
increased the end-to-end latency of live streaming to over 30
seconds, which makes it unattractive for many use cases. Over time,
the industry optimized these delivery methods, bringing the latency
down to 3 seconds.
While the delivery side scaled up, improvements to video transcoding
became a necessity. Viewers watch video streams on a variety of
different devices, connected over different types of networks. Since
upload bandwidth from on-premises locations is often limited, video
transcoding moved to the cloud.
RTMP became the de facto standard for contribution over the public
internet. But there are limitations for the payload to be
transmitted, since RTMP as a media specific protocol only supports
two audio channels and a restricted set of audio and video codecs,
lacking support for newer formats such as HEVC[H.265], VP9[VP9], or
AV1[AV1].
Since RTMP, HLS and DASH rely on TCP, these protocols can only
guarantee acceptable reliability over connections with low RTTs, and
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can not use the bandwidth of network connections to their full extent
due to limitations imposed by congestion control. Notably,
QUIC[I-D.ietf-quic-transport] has been designed to address these
problems with HTTP-based delivery protocols in
HTTP/3[I-D.ietf-quic-http]. Like QUIC, SRT[SRTSRC] uses UDP instead
of the TCP transport protocol, but includes features which assure
more reliable delivery.
1.2. Secure Reliable Transport Protocol
Low latency video transmissions across reliable (usually local) IP
based networks typically take the form of MPEG-TS[ISO13818-1] unicast
or multicast streams using the UDP/RTP protocol, where any packet
loss can be mitigated by enabling forward error correction (FEC).
Achieving the same low latency between sites in different cities,
countries or even continents is more challenging. While it is
possible with satellite links or dedicated MPLS[RFC3031] networks,
these are expensive solutions. The use of public internet
connectivity, while less expensive, imposes significant bandwidth
overhead to achieve the necessary level of packet loss recovery.
Introducing selective packet retransmission (reliable UDP) to recover
from packet loss removes those limitations.
Derived from the UDP-based Data Transfer protocol (UDT), SRT is a
user-level protocol that retains most of the core concepts and
mechanisms while introducing several refinements and enhancements,
including control packet modifications, improved flow control for
handling live streaming, enhanced congestion control, and a mechanism
for encrypting packets.
SRT is a transport protocol that enables the secure, reliable
transport of data across unpredictable networks, such as the
Internet. While any data type can be transferred via SRT, it is
ideal for low latency (sub-second) video streaming. SRT provides
improved bandwidth utilization compared to RTMP, allowing much higher
contribution bitrates over long distance connections.
As packets are streamed from source to destination, SRT detects and
adapts to the real-time network conditions between the two endpoints,
and helps compensate for jitter and bandwidth fluctuations due to
congestion over noisy networks. Its error recovery mechanism
minimizes the packet loss typical of Internet connections.
To achieve low latency streaming, SRT had to address timing issues.
The characteristics of a stream from a source network are completely
changed by transmission over the public internet, which introduces
delays, jitter, and packet loss. This, in turn, leads to problems
with decoding, as the audio and video decoders do not receive packets
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at the expected times. The use of large buffers helps, but latency
is increased.
SRT includes a mechanism that recreates the signal characteristics on
the receiver side, reducing the need for buffering.
Like TCP, SRT employs a listener/caller model. The data flow is bi-
directional and independent of the connection initiation - either the
sender or receiver can operate as listener or caller to initiate a
connection. The protocol provides an internal multiplexing
mechanism, allowing multiple SRT connections to share the same UDP
port, providing access control functionality to identify the caller
on the listener side.
Supporting forward error correction (FEC) and selective packet
retransmission (ARQ), SRT provides the flexibility to use either of
the two mechanisms or both combined, allowing for use cases ranging
from the lowest possible latency to the highest possible reliability.
SRT maintains the ability for fast file transfers introduced in UDT,
and adds support for AES encryption.
2. Conventions and Definitions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP
14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
3. Packet Structure
SRT packets are transmitted in UDP packets [RFC0768]. Every UDP
packet carrying SRT traffic contains an SRT header (immediately after
the UDP header).
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SrcPort | DstPort |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Len | ChkSum |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ SRT Packet +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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Figure 1: SRT packet as UDP payload
SRT has two types of packets distinguished by the Packet Type Flag:
data packet and control packet. The structure of the SRT packet is
shown in Figure 2.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| (Field meaning depends on the packet type) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| (Field meaning depends on the packet type) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- CIF -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Packet Contents |
| (depends on the packet type) +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: SRT packet structure
F (1 bit): Packet Type Flag. The control packet has this flag set
to "1". The data packet has this flag set to "0".
Timestamp (32 bits): The time stamp of the packet in microseconds.
The value is relative to the time the SRT connection was
established. Depending on the transmission mode (Section 4.2),
the field stores the packet send time or the packet origin time.
Destination Socket ID (32 bits): A fixed-width field providing the
SRT socket ID to which a packet should be dispatched. The field
may have the special value "0" when the packet is a connection
request.
3.1. Data Packets
The structure of the SRT data packet is shown in Figure 3.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Packet Sequence Number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|P P|O|K K|R| Message Number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Data +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: data packet structure
Packet Sequence Number (31 bits): The sequential number of the data
packet.
PP (2 bits): Packet Position Flag. This field indicates the
position of the data packet in the message. The value "10b"
(binary) means the first packet of the message. "00b" indicates a
packet in the middle. "01b" designates the last packet. If a
single data packet forms the whole message, the value is "11b".
O (1 bit): Order Flag. Indicates whether the message should be
delivered by the receiver in order (1) or not (0). Certain
restrictions apply depending on the data transmission mode used
(Section 4.2).
KK (2 bits): Key-based Encryption Flag. The flag bits indicate
whether or not data is encrypted. The value "00b" (binary) means
data is not encrypted. "01b" indicates that data is encrypted with
an even key, and "10b" is used for odd key encryption. Refer to
Section 5. The value "11b" is only used in control packets.
R (1 bit): Retransmitted Packet Flag. This flag is clear when a
packet is transmitted the first time. The flag is set to "1" when
a packet is retransmitted.
Message Number (26 bits): The sequential number of consecutive data
packets that form a message (see PP field).
Data (variable length): The payload of the data packet. The length
of the data is the remaining length of the UDP packet.
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3.2. Control Packets
An SRT control packet has the following structure.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Control Type | Subtype |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type-specific Information |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- CIF -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Control Information Field +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 4: control packet structure
Control Type (15 bits): Control Packet Type. The use of these bits
is determined by the control packet type definition. See Table 1.
Subtype (16 bits): This field specifies an additional subtype for
specific packets. See Table 1.
Type-specific Information (32 bits): The use of this field depends
on the particular control packet type. Handshake packets do not
use this field.
Control Information Field (variable length): The use of this field
is defined by the Control Type field of the control packet.
The types of SRT control packets are shown in Table 1. The value
"0x7ffff" is reserved for a user-defined type.
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+-------------------+--------------+---------+---------------+
| Packet Type | Control Type | Subtype | Section |
+===================+==============+=========+===============+
| HANDSHAKE | 0x0000 | 0x0 | Section 3.2.1 |
+-------------------+--------------+---------+---------------+
| KEEPALIVE | 0x0001 | 0x0 | Section 3.2.2 |
+-------------------+--------------+---------+---------------+
| ACK | 0x0002 | 0x0 | Section 3.2.3 |
+-------------------+--------------+---------+---------------+
| NAK (Loss Report) | 0x0003 | 0x0 | Section 3.2.4 |
+-------------------+--------------+---------+---------------+
| SHUTDOWN | 0x0005 | 0x0 | Section 3.2.5 |
+-------------------+--------------+---------+---------------+
| ACKACK | 0x0006 | 0x0 | Section 3.2.6 |
+-------------------+--------------+---------+---------------+
| User-Defined Type | 0x7FFF | - | N/A |
+-------------------+--------------+---------+---------------+
Table 1: SRT Control Packet Types
3.2.1. Handshake
Handshake control packets (Control Type = 0x0000) are used to
exchange peer configurations, to agree on connection parameters, and
to establish a connection.
The Control Information Field (CIF) of a handshake control packet is
shown in Figure 5.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Encryption Field | Extension Field |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Initial Packet Sequence Number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Transmission Unit Size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Maximum Flow Window Size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Handshake Type |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SRT Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SYN Cookie |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ +
| |
+ Peer IP Address +
| |
+ +
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| Extension Type | Extension Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Extension Contents +
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
Figure 5: handshake packet structure
Version (32 bits): A base protocol version number. Currently used
values are 4 and 5. Values greater than 5 are reserved for future
use.
Encryption Field (16 bits): Block cipher family and block size. The
values of this field are described in Table 2.
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+-------+------------------------------+
| Value | Cipher family and block size |
+=======+==============================+
| 0 | No Encryption |
+-------+------------------------------+
| 2 | AES-128 |
+-------+------------------------------+
| 3 | AES-192 |
+-------+------------------------------+
| 4 | AES-256 |
+-------+------------------------------+
Table 2: Handshake Encryption Field
Values
Extension Field (16 bits): This field is message specific extension
related to Handshake Type field. The value must be set to 0
except for the following cases. (1) If the handshake control
packet is the INDUCTION message, this field is sent back by the
Listener. (2) In the case of a CONCLUSION message, this field
value should contain a combination of Extension Type values. For
more details, see Section 4.3.1.
+------------+--------+
| Bitmask | Flag |
+============+========+
| 0x00000001 | HSREQ |
+------------+--------+
| 0x00000002 | KMREQ |
+------------+--------+
| 0x00000004 | CONFIG |
+------------+--------+
Table 3: Handshake
Extension Flags
Initial Packet Sequence Number (32 bits): The sequence number of the
very first data packet to be sent.
Maximum Transmission Unit Size (32 bits): This value is typically
set to 1500, which is the default Maximum Transmission Unit (MTU)
size for Ethernet, but can be less.
Maximum Flow Window Size (32 bits): The value of this field is the
maximum number of data packets allowed to be "in flight"
(i.e. the number of sent packets for which an ACK control packet
has not yet been received).
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Handshake Type (32 bits): This field indicates the handshake packet
type. The possible values are described in Table 4. For more
details refer to Section 4.3.
+------------+----------------+
| Value | Handshake type |
+============+================+
| 0xFFFFFFFD | DONE |
+------------+----------------+
| 0xFFFFFFFE | AGREEMENT |
+------------+----------------+
| 0xFFFFFFFF | CONCLUSION |
+------------+----------------+
| 0x00000000 | WAVEHAND |
+------------+----------------+
| 0x00000001 | INDUCTION |
+------------+----------------+
Table 4: Handshake Type
SRT Socket ID (32 bits): This field holds the ID of the source SRT
socket from which a handshake packet is issued.
SYN Cookie (32 bits): Randomized value for processing a handshake.
The value of this field is specified by the handshake message
type. See Section 4.3.
Peer IP Address (128 bits): The sender's IPv4 or IPv6 address. The
value consists of four 32-bit fields. In the case of IPv4
addresses, fields 2, 3 and 4 are padded with zeroes.
Extension Type (16 bits): The value of this field is used to process
an integrated handshake. There are two extensions: Handshake
Extension Message (Section 3.2.1.1) and Key Material Exchange
(Section 3.2.1.2). Each extension can have a pair of request and
response types.
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+-------+--------------------+-------------------+
| Value | Extension Type | HS Extension Flag |
+=======+====================+===================+
| 1 | SRT_CMD_HSREQ | HSREQ |
+-------+--------------------+-------------------+
| 2 | SRT_CMD_HSRSP | HSREQ |
+-------+--------------------+-------------------+
| 3 | SRT_CMD_KMREQ | KMREQ |
+-------+--------------------+-------------------+
| 4 | SRT_CMD_KMRSP | KMREQ |
+-------+--------------------+-------------------+
| 5 | SRT_CMD_SID | CONFIG |
+-------+--------------------+-------------------+
| 6 | SRT_CMD_CONGESTION | CONFIG |
+-------+--------------------+-------------------+
| 7 | SRT_CMD_FILTER | CONFIG |
+-------+--------------------+-------------------+
| 8 | SRT_CMD_GROUP | CONFIG |
+-------+--------------------+-------------------+
Table 5: Handshake Extension Type values
Extension Length (16 bits): The length of the Extension Contents
field.
Extension Contents (variable length): The payload of the extension.
3.2.1.1. Handshake Extension Message
In a Handshake Extension, the value of the Extension Field of the
handshake control packet is defined as 1 for a Handshake Extension
request, and 2 for a Handshake Extension response.
The Extension Contents field of a Handshake Extension Message is
structured as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SRT Version |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SRT Flags |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Receiver TSBPD Delay | Sender TSBPD Delay |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: Handshake Extension Message structure
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SRT Version (32 bits): SRT library version.
SRT Flags (32 bits): SRT configuration flags:
+------------+---------------+
| Bitmask | Flag |
+============+===============+
| 0x00000001 | TSBPDSND |
+------------+---------------+
| 0x00000002 | TSBPDRCV |
+------------+---------------+
| 0x00000004 | CRYPT |
+------------+---------------+
| 0x00000008 | TLPKTDROP |
+------------+---------------+
| 0x00000010 | PERIODICNAK |
+------------+---------------+
| 0x00000020 | REXMITFLG |
+------------+---------------+
| 0x00000040 | STREAM |
+------------+---------------+
| 0x00000080 | PACKET_FILTER |
+------------+---------------+
Table 6: Handshake
Extension Message Flags
Receiver TSBPD Delay (16 bits): TimeStamp-Based Packet Delivery
(TSBPD) Delay of the receiver. Refer to Section 4.5.
Sender TSBPD Delay (16 bits): TSBPD of the sender. Refer to
Section 4.5.
3.2.1.2. Key Material Exchange
The Key Material Exchange portion of a Handshake packet has both
request and response type extensions. The value of a request is 3,
and the response value is 4.
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|S| V | PT | Sign | Resv | KK|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| KEKI |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Cipher | Auth | SE | SLen | KLen |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Salt |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Wrapped Key +
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: Key Material Extension structure
S ( ): 1 bit. Value: {0} This is a fixed-width field that is a
remnant from the header of a previous design.
Version (V): 3 bits. Value: {1} This is a fixed-width field that
indicates the SRT version: - 1: initial version
Packet Type (PT): 4 bits. Value: {2} This is a fixed-width field
that indicates the Packet Type: - 0: Reserved - 1: MSmsg - 2:
KMmsg - 7: Reserved to discriminate MPEG-TS packet (0x47=sync
byte)
Signature (Sign): 16 bits. Value: {0x2029} This is a fixed-width
field that contains the signature 'HAI' encoded as a PnP Vendor ID
([PNPID]) (in big endian order)
Reserved (Resv): 6 bits. Value: {0} This is a fixed-width field
reserved for flag extension or other usage.
Key-based Data Encryption (KK): 2 bits. This is a fixed-width field
that indicates whether or not data is encrypted: - 00b: not
encrypted (data packets only) - 01b: even key - 10b: odd key
- 11b: even and odd keys
Key Encryption Key Index (KEKI): 32 bits. Value: {0} This is a
fixed-width field for specifying the KEK index (big endian order)
- 0: Default stream associated key (stream/system default) -
1..255: Reserved for manually indexed keys
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Cipher ( ): 8 bits. Value: {0..2} This is a fixed-width field for
specifying encryption cipher and mode: - 0: None or KEKI indexed
crypto context - 1: AES-ECB (not supported in SRT) - 2: AES-CTR
[SP800-38A]
Authentication (Auth): 8 bits. Value: {0} This is a fixed-width
field for specifying a message authentication code algorithm: - 0:
None or KEKI indexed crypto context
Stream Encapsulation (SE): 8 bits. Value: {2} This is a fixed-width
field for describing the stream encapsulation: - 0: Unspecified or
KEKI indexed crypto context - 1: MPEG-TS/UDP - 2: MPEG-TS/SRT
Reserved (Resv1): 8 bits. Value: {0} This is a fixed-width field
reserved for future use.
Reserved (Resv2): 16 bits. Value: {0} This is a fixed-width field
reserved for future use.
Slen/4 ( ): 4 bits. Value: {0..255} This is a fixed-width field for
specifying salt length in bytes divided by 4. Can be zero if no
salt/IV present
Klen/4 ( ): 8 bits. Value: {4,6,8} This is a fixed-width field for
specifying SEK length in bytes divided by 4. Size of one key even
if two keys present.
Salt (Slen): Slen*8 bits. Value: { } This is a variable-width field
for specifying a salt key
Wrap ( ): (64+n * Klen * 8) bits. Value: { } This is a variable-
width field for specifying Wrapped key(s), where n = 1 or 2 NOTE
1: n = (KK + 1)/2 NOTE 2: size in bytes = (((KK+1/2) * Klen) + 8)
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
+ Integrity Check Vector (ICV) +
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| xSEK |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| oSEK |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
Figure 8: Unwrapped key structure
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ICV (64 bits): 64-bit Integrity Check Vector(AES key wrap
integrity).
xSEK (variable length): This field identifies an odd or even SEK.
If both keys are present, then this field is eSEK (even key) and
the next one is the odd key. The length of this field is
calculated by KLen * 4 * 8.
oSEK (variable length): This field is present only when the message
carries the two SEKs.
3.2.2. Keep-Alive
Keep-Alive control packets are sent after a certain timeout from the
last time any packet (Control or Data) was sent. The purpose of this
control packet is to notify the peer to keep the connection open when
no data exchange is taking place.
The default timeout for a Keep-Alive packet to be sent is 1 second.
An SRT Keep-Alive packet is formatted as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Control Type | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type-specific Information |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Time Stamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- CIF -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CIF (none) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 9: Keep-Alive structure
Packet Type ( ): 1 bit. Value: 1 The type value of a Keep-Alive
control packet is "1".
Control Type ( ): 15 bits. Value: KEEPALIVE{1} This is a fixed-
width field used to indicate message type
Reserved ( ): 16 bits. Value: ??? This is a fixed-width field
reserved for future use.
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Type-specific Information: This field is reserved for future
definition.
Time Stamp (TS): 32 bits. Value: ??? This is a fixed-width field
usually containing the time (in microseconds) when a packet was
sent, although the real interpretation may vary depending on the
type.
Destination Socket ID (DestSockID): 32 bits. Value: ??? This is a
fixed-width field providing the socket ID to which a packet should
be dispatched, although it may have the special value 0 when the
packet is a connection request.
Control Information Field (CIF): n bits. Value: {none} This field
must not appear in Keep-Alive control packets.
3.2.3. ACK (Acknowledgement)
Acknowledgement control packets are used to provide delivery status
of data packets. These packets may also carry some additional
information from the receiver like RTT, bandwidth, receiving speed,
etc. The CIF portion of the ACK control packet is expanded as
follows:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Control Type | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type-specific Information |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Time Stamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- CIF -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Last Acknowledged Packet Sequence Number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTT |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTT variance |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Available Buffer Size |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Packets Receiving Rate |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Estimated Link Capacity |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Receiving Rate |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 10: ACK control packet
Type-specific Information (32 bits): The time-specific Information
(Figure 4) of the ACK packet stores the sequential number of the
full ACK packet starting from 1.
Last Acknowledged Packet Sequence Number (32 bits): The sequence
number of the last acknowledged data packet +1.
RTT (32 bits): RTT value (in microseconds) estimated by the receiver
based on the previous ACK-ACKACK packet exchange.
RTT variance (32 bits): The variance of the RTT estimation (in
microseconds).
Available Buffer Size (32 bits): Available size of the receiver's
buffer (in packets).
Packets Receiving Rate (32 bits): The rate at which packets are
being received (in packets per second).
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Estimated Link Capacity (32 bits): Estimated bandwidth of the link
(in packets per second).
Receiving Rate (32 bits): Estimated receiving rate (in bytes per
second).
There are several types of ACK packets:
* A Full ACK control packet is sent every 10 ms and has all the
fields of Figure 10.
* A Lite ACK control packet includes only the Last Acknowledged
Packet Sequence Number field. The Type-specific Information field
should be set to 0.
* A Small ACK includes the fields up to and including the Available
Buffer Size field. The Type-specific Information field should be
set to 0.
The sender only acknowledges the receipt of Full ACK packets (see
ACKACK).
The Lite ACK and Small ACK packets are used in cases when the
receiver should acknowledge received data packets more often than
every 10 ms. This is usually needed at high data rates. It is up to
the receiver to decide the condition and the type of ACK packet to
send (Lite or Small). The recommendation is to send a Lite ACK for
every 64 packets received.
3.2.4. NAK (Loss Report)
Negative acknowledgement (NAK) control packets are used to signal
failed data packet deliveries. The receiver notifies the sender
about lost data packets by sending a NAK packet that contains a list
of sequence numbers for those lost packets.
An SRT NAK packet is formatted as follows:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Control Type | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type-specific Information |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Time Stamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Lost packet sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| List of lost packets |
+-+-+-+-+-+-+-+-+-+-+-+- CIF (Loss List) -+-+-+-+-+-+-+-+-+-+-+-+
|0| Up to |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Lost packet sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 11: NAK control packet
Control Type: The type value of a NAK control packet is "3".
Type-specific Information: This field is reserved for future
definition.
Control Information Field (CIF): A single value or a list of lost
packets sequence numbers. See packet sequence number coding in
Appendix A.
3.2.5. Shutdown
Shutdown control packets are used to initiate the closing of an SRT
connection.
An SRT SHUTDOWN Control packet is formatted as follows:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Control Type | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type-specific Information |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Time Stamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- CIF -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| None |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 12: SHUTDOWN control packet
Control Type: The type value of Shutdown control packet is "5".
Type-specific Information: This field is reserved for future
definition.
Control Information Field: This field must not appear in shutdown
control packets.
3.2.6. ACKACK
ACKACK control packets are sent to acknowledge the reception of a
Full ACK, and are used in the calculation of RTT by the receiver.
An SRT ACKACK Control packet is formatted as follows:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+- SRT Header +-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Control Type | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Type-specific Information |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Time Stamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Destination Socket ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- CIF -+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| None |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 13: ACKACK control packet
Control Type: The type value of ACKACK control packet is "6".
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Type-specific Information: ACK Sequence Number. This field is used
for the sequence number of the ACK packet being acknowledged.
Control Information Field: This field must not appear in ACKACK
control packets.
4. SRT Data Transmission and Control
This section describes key concepts related to the handling of
control and data packets during the transmission process.
After the handshake and exchange of capabilities is completed, packet
data can be sent and received over the established connection. To
fully utilize the features of low latency and error recovery provided
by SRT, the sender and receiver MUST handle control packets, timers,
and buffers for the connection as specified in this section.
4.1. Stream Multiplexing
Multiple SRT sockets may share the same UDP socket so that the
packets received to this UDP socket will be correctly dispatched to
the SRT socket they are currently destined.
During the handshake, the parties exchange their SRT Socket IDs.
These IDs are then used in the Destination Socket ID field of every
control and data packet (see Section 3).
4.2. Data Transmission Modes
SRT has been mainly created for Live Streaming and therefore its main
and default transmission mode is "live". SRT supports, however, the
modes that the original UDT library supported, that is, buffer and
message transmission.
4.2.1. Message Mode
When the STREAM flag of the handshake Extension Message
Section 3.2.1.1 is set to 0, the protocol operates in Message mode,
characterized as follows:
* Every packet has its own Packet Sequence Number.
* One or several consecutive SRT Data packets can form a message.
* All the packets belonging to the same message have a similar
message number set in the Message Number field.
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The first packet of a message has the first bit of the Packet
Position Flags (Section 3.1) set to 1. The last packet of the
message has the second bit of the Packet Position Flags set to 1.
Thus, a PP equal to "11b" indicates a packet that forms the whole
message. A PP equal to "00b" indicates a packet that belongs to the
inner part of the message.
The concept of the message in SRT comes from UDT ([GHG04b]). In this
mode a single sending instruction passes exactly one piece of data
that has boundaries (a message). This message may span across
multiple UDP packets (and multiple SRT data packets). The only size
limitation is that it shall fit as a whole in the buffers of the
sender and the receiver. Although internally all operations (e.g.
ACK, NAK) on data packets are performed independently, an application
MUST send and receive the whole message. Until the message is
complete (all packets are received) the application will not be
allowed to read it.
When the Order Flag of a Data packet is set to 1, this imposes a
sequential reading order on messages. An Order Flag set to 0 allows
an application to read messages that are already fully available,
before any preceding messages that may have some packets missing.
4.2.2. Live Mode
Live mode is a special type of message mode where only data packets
with their PP field set to "11b" are allowed.
Additionally Timestamp Based Packet Delivery (TSBPD) (Section 4.5)
and Too-Late Packet Drop (Section 4.6) mechanisms are used in this
mode.
4.2.3. Buffer Mode
Buffer mode is negotiated during the Handshake by setting the STREAM
flag of the handshake Extension Message Flags to 1.
In this mode consecutive packets form one continuous stream that can
be read, with portions of any size.
4.3. Handshake Messages
SRT is a connection protocol. It embraces the concepts of
"connection" and "session". The UDP system protocol is used by SRT
for sending data and control packets.
An SRT connection is characterized by the fact that it is:
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* first engaged by a handshake process;
* maintained as long as any packets are being exchanged in a timely
manner;
* considered closed when a party receives the appropriate close
command from its peer (connection closed by the foreign host), or
when it receives no packets at all for some predefined time
(connection broken on timeout).
SRT supports two connection configurations:
1. Caller-Listener, where one side waits for the other to initiate a
connection
2. Rendezvous, where both sides attempt to initiate a connection
The handshake is performed between two parties: "Initiator" and
"Responder":
* Initiator starts the extended SRT handshake process and sends
appropriate SRT extended handshake requests.
* Responder expects the SRT extended handshake requests to be sent
by the Initiator and sends SRT extended handshake responses back.
There are two basic types of SRT handshake extensions that are
exchanged in the handshake:
* Handshake Extension Message exchanges the basic SRT information;
* Key Material Exchange exchanges the wrapped stream encryption key
(used only if encryption is requested).
* Stream ID extension exchanges some stream-specific information
that can be used by the application to identify the incoming
stream connection.
The Initiator and Responder roles are assigned depending on the
connection mode.
For Caller-Listener connections: the Caller is the Initiator, the
Listener is the Responder. For Rendezvous connections: the Initiator
and Responder roles are assigned based on the initial data
interchange during the handshake.
The Handshake Type field in the Handshake Structure (see Figure 5)
indicates the handshake message type.
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Caller-Listener handshake exchange has the following order of
Handshake Types:
1. Caller to Listener: INDUCTION
2. Listener to Caller: INDUCTION (reports cookie)
3. Caller to Listener: CONCLUSION (uses previously returned cookie)
4. Listener to Caller: CONCLUSION (confirms connection established)
Rendezvous handshake exchange has the following order of Handshake
Types:
1. After starting the connection: WAVEAHAND.
2. After receiving the above message from the peer: CONCLUSION.
3. After receiving the above message from the peer: AGREEMENT.
When a connection process has failed before either party can send the
CONCLUSION handshake, the Handshake Type field will contain the
appropriate error value for the rejected connection. See the list of
error codes in Table 7.
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+------+----------------+-----------------------------------------+
| Code | Error | Description |
+======+================+=========================================+
| 1000 | REJ_UNKNOWN | Unknown reason |
+------+----------------+-----------------------------------------+
| 1001 | REJ_SYSTEM | System function error |
+------+----------------+-----------------------------------------+
| 1002 | REJ_PEER | Rejected by peer |
+------+----------------+-----------------------------------------+
| 1003 | REJ_RESOURCE | Resource allocation problem |
+------+----------------+-----------------------------------------+
| 1004 | REJ_ROGUE | incorrect data in handshake |
+------+----------------+-----------------------------------------+
| 1005 | REJ_BACKLOG | listener's backlog exceeded |
+------+----------------+-----------------------------------------+
| 1006 | REJ_IPE | internal program error |
+------+----------------+-----------------------------------------+
| 1007 | REJ_CLOSE | socket is closing |
+------+----------------+-----------------------------------------+
| 1008 | REJ_VERSION | peer is older version than agent's min |
+------+----------------+-----------------------------------------+
| 1009 | REJ_RDVCOOKIE | rendezvous cookie collision |
+------+----------------+-----------------------------------------+
| 1010 | REJ_BADSECRET | wrong password |
+------+----------------+-----------------------------------------+
| 1011 | REJ_UNSECURE | password required or unexpected |
+------+----------------+-----------------------------------------+
| 1012 | REJ_MESSAGEAPI | Stream flag collision |
+------+----------------+-----------------------------------------+
| 1013 | REJ_CONGESTION | incompatible congestion-controller type |
+------+----------------+-----------------------------------------+
| 1014 | REJ_FILTER | incompatible packet filter |
+------+----------------+-----------------------------------------+
| 1015 | REJ_GROUP | incompatible group |
+------+----------------+-----------------------------------------+
Table 7: Handshake Rejection Reason Codes
The specification of the cipher family and block size is decided by
the Sender. When the transmission is bidirectional, this value must
be agreed upon at the outset because when both are set the Responder
wins. For Caller-Listener connections it is reasonable to set this
value on the Listener only. In the case of Rendezvous the only
reasonable approach is to decide upon the correct value from the
different sources and to set it on both parties (note that *AES-128*
is the default).
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4.3.1. Caller-Listener Handshake
This section describes the handshaking process where a Listener is
waiting for an incoming Handshake request on a bound UDP port from a
Caller. The process has two phases: induction and conclusion.
4.3.1.1. The Induction Phase
The Caller begins by sending the INDUCTION handshake, which contains
the following (significant) fields:
* Version: must always be 4
* Encryption Field: 0
* Extension Field: 2
* Handshake Type: INDUCTION
* SRT Socket ID: SRT Socket ID of the Caller
* SYN Cookie: 0
The Destination Socket ID of the SRT packet header in this message is
0, which is interpreted as a connection request.
The handshake version number is set to 4 in this initial handshake.
This is due to the initial design of SRT that was to be compliant
with the UDT protocol ([GHG04b]) on which it is based.
This phase serves only to set a cookie on the Listener so that it
doesn't allocate resources, thus mitigating a potential DoS attack
that might be perpetrated by flooding the Listener with handshake
commands.
The Listener responds with the following:
* Version: 5
* Encryption Field: Advertised cipher family and block size.
* Extension Field: SRT magic code 0x4A17
* Handshake Type: INDUCTION
* SRT Socket ID: Socket ID of the Listener
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* SYN Cookie: a cookie that is crafted based on host, port and
current time with 1 minute accuracy
At this point the Listener still doesn't know if the Caller is SRT or
UDT, and it responds with the same set of values regardless of
whether the Caller is SRT or UDT.
If the party is SRT, it does interpret the values in Version and
Extension Field. If it receives the value 5 in Version, it
understands that it comes from an SRT party, so it knows that it
should prepare the proper handshake messages phase. It also checks
the following:
* whether the Extension Flags contains the magic value 0x4A17;
otherwise the connection is rejected. This is a contingency for
the case where someone who, in an attempt to extend UDT
independently, increases the Version value to 5 and tries to test
it against SRT.
* whether the Encryption Flags contain a non-zero value, which is
interpreted as an advertised cipher family and block size.
A legacy UDT party completely ignores the values reported in Version
and Handshake Type. It is, however, interested in the SYN Cookie
value, as this must be passed to the next phase. It does interpret
these fields, but only in the "conclusion" message.
4.3.1.2. The Conclusion Phase
Once the Caller gets the SYN cookie from the Listener, it sends the
CONCLUSION handshake to the Listener.
The following values are set by the compliant caller:
* Version: 5
* Handshake Type: CONCLUSION
* SRT Socket ID: Socket ID of the Caller
* SYN Cookie: the cookie previously received in the induction phase
The Destination Socket ID in this message is the socket ID that was
previously received in the induction phase in the SRT Socket ID field
of the handshake structure.
* Encryption Flags: advertised cipher family and block size.
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* Extension Flags: A set of flags that define the extensions
provided in the handshake.
The Listener responds with the same values shown above, without the
cookie (which is not needed here), as well as the extensions for HS
Version 5 (which will probably be exactly the same).
There is not any "negotiation" here. If the values passed in the
handshake are in any way not acceptable by the other side, the
connection will be rejected. The only case when the Listener can
have precedence over the Caller is the advertised Cipher Family and
Block Size (Table 2) in the Encryption Field of the Handshake.
The value for latency is always agreed to be the greater of those
reported by each party.
4.3.2. Rendezvous Handshake
The Rendezvous process uses a state machine. It is slightly
different from UDT Rendezvous handshake [GHG04b], although it is
still based on the same message request types.
Both parties start with WAVEAHAND and use the Version value of 5.
Legacy Version 4 clients do not look at the Version value, whereas
Version 5 clients can detect version 5. The parties only continue
with the Version 5 Rendezvous process when Version is set to 5 for
both. Otherwise the process continues exclusively according to
Version 4 rules [GHG04b].
With Version 5 Rendezvous, both parties create a cookie for a process
called the "cookie contest". This is necessary for the assignment of
Initiator and Responder roles. Each party generates a cookie value
(a 32-bit number) based on the host, port, and current time with 1
minute accuracy. This value is scrambled using an MD5 sum
calculation. The cookie values are then compared with one another.
Since it is impossible to have two sockets on the same machine bound
to the same NIC and port and operating independently, it is virtually
impossible that the parties will generate identical cookies.
However, this situation may occur if an application tries to "connect
to itself" - that is, either connects to a local IP address, when the
socket is bound to INADDR_ANY, or to the same IP address to which the
socket was bound. If the cookies are identical (for any reason), the
connection will not be made until new, unique cookies are generated
(after a delay of up to one minute). In the case of an application
"connecting to itself", the cookies will always be identical, and so
the connection will never be established.
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When one party's cookie value is greater than its peer's, it wins the
cookie contest and becomes Initiator (the other party becomes the
Responder).
At this point there are two possible "handshake flows": _serial_ and
_parallel_.
4.3.2.1. Serial Handshake Flow
In the serial handshake flow, one party is always first, and the
other follows. That is, while both parties are repeatedly sending
WAVEAHAND messages, at some point one party - let's say Alice - will
find she has received a WAVEAHAND message before she can send her
next one, so she sends a CONCLUSION message in response. Meantime,
Bob (Alice's peer) has missed Alice's WAVEAHAND messages, so that
Alice's CONCLUSION is the first message Bob has received from her.
This process can be described easily as a series of exchanges between
the first and following parties (Alice and Bob, respectively):
1. Initially, both parties are in the _waving_ state. Alice sends a
handshake message to Bob:
* Version: 5
* Type: Extension field: 0, Encryption field: advertised
"PBKEYLEN".
* Handshake Type: WAVEAHAND
* SRT Socket ID: Alice's socket ID
* SYN Cookie: Created based on host/port and current time.
While Alice doesn't yet know if she is sending this message to a
Version 4 or Version 5 peer, the values from these fields would not
be interpreted by the Version 4 peer when the Handshake Type is
WAVEAHAND.
1. Bob receives Alice's WAVEAHAND message, switches to the
"attention" state. Since Bob now knows Alice's cookie, he
performs a "cookie contest" (compares both cookie values). If
Bob's cookie is greater than Alice's, he will become the
Initiator. Otherwise, he will become the Responder.
The resolution of the Handshake Role (Initiator or Responder) is
essential for further processing.
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Then Bob responds:
* Version: 5
* Extension field: appropriate flags if Initiator, otherwise 0
* Encryption field: advertised PBKEYLEN
* Handshake Type: CONCLUSION
If Bob is the Initiator and encryption is on, he will use either his
own cipher family and block size or the one received from Alice (if
she has advertised those values).
1. Alice receives Bob's CONCLUSION message. While at this point she
also performs the "cookie contest", the outcome will be the same.
She switches to the "fine" state, and sends:
* Version: 5
* Appropriate extension flags and encryption flags
* Handshake Type: CONCLUSION
Both parties always send extension flags at this point, which will
contain HSREQ if the message comes from an Initiator, or HSRSP if it
comes from a Responder. If the Initiator has received a previous
message from the Responder containing an advertised cipher family and
block size in the encryption flags field, it will be used as the key
length for key generation sent next in the KMREQ extension.
1. Bob receives Alice's CONCLUSION message, and then does one of the
following (depending on Bob's role):
* If Bob is the Initiator (Alice's message contains HSRSP), he:
- switches to the "*connected" state
- sends Alice a message with Handshake Type AGREEMENT, but
containing no SRT extensions (Extension Flags field should
be 0)
* If Bob is the Responder (Alice's message contains HSREQ), he:
- switches to "initiated" state
- sends Alice a message with Handshake Type CONCLUSION that
also contains extensions with HSRSP
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o awaits a confirmation from Alice that she is also
connected (preferably by AGREEMENT message)
2. Alice receives the above message, enters into the "connected"
state, and then does one of the following (depending on Alice's
role):
* If Alice is the Initiator (received CONCLUSION with HSRSP),
she sends Bob a message with Handshake Type = URQ_AGREEMENT.
* If Alice is the Responder, the received message has Handshake
Type AGREEMENT and in response she does nothing.
3. At this point, if Bob was Initiator, he is connected already. If
he was a Responder, he should receive the above AGREEMENT
message, after which he switches to the "connected" state. In
the case where the UDP packet with the agreement message gets
lost, Bob will still enter the _connected_ state once he receives
anything else from Alice. If Bob is going to send, however, he
has to continue sending the same CONCLUSION until he gets the
confirmation from Alice.
4.3.2.2. Parallel Handshake Flow
The chances of the parallel handshake flow are very low, but still it
may occur if the handshake messages with WAVEAHAND are sent and
received by both peers at precisely the same time.
The resulting flow is very much like Bob's behavior in the serial
handshake flow, but for both parties. Alice and Bob will go through
the same state transitions:
Waving -> Attention -> Initiated -> Connected
In the Attention state they know each other's cookies, so they can
assign roles. In contrast to serial flows, which are mostly based on
request-response cycles, here everything happens completely
asynchronously: the state switches upon reception of a particular
handshake message with appropriate contents (the Initiator must
attach the HSREQ extension, and Responder must attach the "HSRSP"
extension).
Here's how the parallel handshake flow works, based on roles:
Initiator:
1. Waving
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* Receives WAVEAHAND message
* Switches to Attention
* Sends CONCLUSION + HSREQ
2. Attention
* Receives CONCLUSION message, which:
- contains no extensions:
o switches to Initiated, still sends URQ_CONCLUSION +
HSREQ
- contains "HSRSP" extension:
o switches to Connected, sends AGREEMENT
3. Initiated
* Receives CONCLUSION message, which:
- Contains no extensions:
o REMAINS IN THIS STATE, still sends URQ_CONCLUSION +
HSREQ
- contains "HSRSP" extension:
o switches to Connected, sends AGREEMENT
4. Connected
* May receive CONCLUSION and respond with AGREEMENT, but
normally by now it should already have received payload
packets.
Responder:
1. Waving
* Receives WAVEAHAND message
* Switches to Attention
* Sends CONCLUSION message (with no extensions)
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2. Attention
* Receives CONCLUSION message with HSREQ This message might
contain no extensions, in which case the party shall simply
send the empty CONCLUSION message, as before, and remain in
this state.
* Switches to Initiated and sends CONCLUSION message with HSRSP
3. Initiated
* Receives:
- CONCLUSION message with HSREQ
o responds with CONCLUSION with HSRSP and remains in this
state
- AGREEMENT message
o responds with AGREEMENT and switches to Connected
- Payload packet
o responds with AGREEMENT and switches to Connected
4. Connected
* Is not expecting to receive any handshake messages anymore.
The AGREEMENT message is always sent only once or per every
final CONCLUSION message.
Note that any of these packets may be missing, and the sending party
will never become aware. The missing packet problem is resolved this
way:
1. If the Responder misses the CONCLUSION + HSREQ message, it simply
continues sending empty CONCLUSION messages. Only upon reception
of CONCLUSION + HSREQ does it respond with CONCLUSION + HSRSP.
2. If the Initiator misses the CONCLUSION + HSRSP response from the
Responder, it continues sending CONCLUSION + HSREQ. The
Responder must always respond with CONCLUSION + HSRSP when the
Initiator sends CONCLUSION + HSREQ, even if it has already
received and interpreted it.
3. When the Initiator switches to the Connected state it responds
with a AGREEMENT message, which may be missed by the Responder.
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Nonetheless, the Initiator may start sending data packets because
it considers itself connected - it doesn't know that the
Responder has not yet switched to the Connected state. Therefore
it is exceptionally allowed that when the Responder is in the
Initiated state and receives a data packet (or any control packet
that is normally sent only between connected parties) over this
connection, it may switch to the Connected state just as if it
had received a AGREEMENT message.
4. If the the Initiator has already switched to the Connected state
it will not bother the Responder with any more handshake
messages. But the Responder may be completely unaware of that
(having missed the AGREEMENT message from the Initiator).
Therefore it doesn't exit the connecting state, which means that
it continues sending CONCLUSION + HSRSP messages until it
receives any packet that will make it switch to the Connected
state (normally AGREEMENT). Only then does it exit the
connecting state and the application can start transmission.
4.4. SRT Buffer Latency
The SRT sender and receiver have buffers to store packets.
On the sender, latency is the time that SRT holds a packet to give it
a chance to be delivered successfully while maintaining the rate of
the sender at the receiver. If an acknowledgement (ACK) is missing
or late for more than the configured latency, the packet is dropped
from the sender buffer. A packet can be retransmitted as long as it
remains in the buffer for the duration of the latency window. On the
receiver, packets are delivered to an application from a buffer after
the latency interval has passed. This helps to recover from
potential packet losses. See sections Section 4.5, Section 4.6 for
details.
Latency is a value (specified in milliseconds) that can cover the
time to transmit hundreds or even thousands of packets at high
bitrate. Latency can be thought of as a window that slides over
time, during which a number of activities take place, such as the
reporting of acknowledged packets (ACKs) (Section 4.8.1) and
unacknowledged packets (NAKs)(Section 4.8.2).
Latency is configured through the exchange of capabilities during the
extended handshake process between initiator and responder. The
Handshake Extension Message (Section 3.2.1.1) has TSBPD delay
information (in milliseconds) from the SRT receiver and sender. The
latency for a connection will be established as the maximum value of
latencies proposed by the initiator and responder.
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4.5. Timestamp Based Packet Delivery
The goal of the SRT Timestamp Based Packet Delivery (TSBPD) mechanism
is to reproduce the output of the sending application (e.g., encoder)
at the input of the receiving application (e.g., decoder) in live
data transmission mode (see Section 4.2). It attempts to reproduce
the timing of packets committed by the sending application to the SRT
sender. This allows packets to be scheduled for delivery by the SRT
receiver, making them ready to be read by the receiving application
(see Figure 14).
The SRT receiver, using the timestamp of the SRT data packet header,
delivers packets to a receiving application with a fixed minimum
delay from the time the packet was scheduled for sending on the SRT
sender side. Basically, the sender timestamp in the received packet
is adjusted to the receiver's local time (compensating for the time
drift or different time zones) before releasing the packet to the
application. Packets can be withheld by the SRT receiver for a
configured receiver delay. A higher delay can accommodate a larger
uniform packet drop rate, or a larger packet burst drop. Packets
received after their "play time" are dropped if the Too-Late Packet
Drop feature is enabled (see Section 4.6).
The packet timestamp (in microseconds) is relative to the SRT
connection creation time. Packets are inserted based on the sequence
number in the header field. The origin time (in microseconds) of the
packet is already sampled when a packet is first submitted by the
application to the SRT sender. The TSBPD feature uses this time to
stamp the packet for first transmission and any subsequent
retransmission. This timestamp and the configured SRT latency
(Section 4.4) control the recovery buffer size and the instant that
packets are delivered at the destination (the aforementioned "play
time" which is decided by adding the timestamp to the configured
latency).
It is worth mentioning that the use of the packet sending time to
stamp the packets is inappropriate for the TSBPD feature, since a new
time (current sending time) is used for retransmitted packets,
putting them out of order when inserted at their proper place in the
stream.
Figure 14 illustrates the key latency points during the packet
transmission with the TSBPD feature enabled.
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| Sending | | |
| Delay | ~RTT/2 | SRT Latency |
|<--------->|<------------>|<----------------->|
| | | |
| | | |
| | | |
___ Scheduled Sent Received Scheduled
/ for sending | | for delivery
Packet | | | |
State | | | |
| | | |
| | | |
----------------------------------------------------->
Time
Figure 14: Key Latency Points during the Packet Transmission
The main packet states shown in Figure 14 are the following:
* "Scheduled for sending": the packet is committed by the sending
application, stamped and ready to be sent;
* "Sent": the packet is passed to the UDP socket and sent;
* "Received": the packet is received and read from the UDP socket;
* "Scheduled for delivery": the packet is scheduled for the delivery
and ready to be read by the receiving application.
It is worth noting that the round-trip time (RTT) of an SRT link may
vary in time. However the actual end-to-end latency on the link
becomes fixed and is approximately equal to (RTT_0/2 + SRT Latency)
once the SRT handshake exchange happens, where RTT_0 is the actual
value of the round-trip time during the SRT handshake exchange (the
value of the round-trip time once the SRT connection has been
established).
The value of sending delay depends on the hardware performance.
Usually it is relatively small (several microseconds) in contrast to
RTT_0/2 and SRT latency which are measured in milliseconds.
4.5.1. Packet Delivery Time
Packet delivery time is the moment, estimated by the receiver, when a
packet should be delivered to the upstream application. The
calculation of packet delivery time (PktTsbpdTime) is performed upon
receiving a data packet according to the following formula:
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PktTsbpdTime = TsbpdTimeBase + PKT_TIMESTAMP + TsbpdDelay + Drift
where
* TsbpdTimeBase is the time base that reflects the time difference
between local clock of the receiver and the clock used by the
sender to timestamp packets being sent (see Section 4.5.1.1);
* PKT_TIMESTAMP is the data packet timestamp, in microseconds;
* TsbpdDelay is the receiver's buffer delay (or receiver's buffer
latency, or SRT Latency). This is the time, in milliseconds, that
SRT holds a packet from the moment it has been received till the
time it should be delivered to the upstream application;
* Drift is the time drift used to adjust the fluctuations between
sender and receiver clock, in microseconds.
SRT Latency (TsbpdDelay) should be a buffer time large enough to
cover the unexpectedly extended RTT time, and the time needed to
retransmit the lost packet. The value of minimum TsbpdDelay is
negotiated during the SRT handshake exchange and is equal to 120
milliseconds. The recommended value of TsbpdDelay is 3-4 times RTT.
it is worth noting that TsbpdDelay limits the number of packet
retransmissions to a certain extent making impossible to retransmit
packets endlessly. This is important for live data transmission.
4.5.1.1. TSBPD Time Base Calculation
The initial value of TSBPD time base (TsbpdTimeBase) is calculated at
the moment of the second handshake request is received as follows:
TsbpdTimeBase = T_NOW - HSREQ_TIMESTAMP
where T_NOW is the current time according to the receiver clock;
HSREQ_TIMESTAMP is the handshake packet timestamp, in microseconds.
The value of TsbpdTimeBase is approximately equal to the initial one-
way delay of the link RTT_0/2, where RTT_0 is the actual value of the
round-trip time during the SRT handshake exchange.
During the transmission process, the value of TSBPD time base may be
adjusted in two cases:
1. During the TSBPD wrapping period.
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The TSBPD wrapping period happens every 01:11:35 hours. This time
corresponds to the maximum timestamp value of a packet
(MAX_TIMESTAMP). MAX_TIMESTAMP is equal to 0xFFFFFFFF, or the
maximum value of 32-bit unsigned integer, in microseconds
(Section 3). The TSBPD wrapping period starts 30 seconds before
reaching the maximum timestamp value of a packet and ends once the
packet with timestamp within (30, 60) seconds interval is delivered
(read from the buffer). The updated value of TsbpdTimeBase will be
recalculated as follows:
TsbpdTimeBase = TsbpdTimeBase + MAX_TIMESTAMP + 1
1. By drift tracer. See Section 4.7 for details.
4.6. Too-Late Packet Drop
The Too-Late Packet Drop (TLPKTDROP) mechanism allows the sender to
drop packets that have no chance to be delivered in time, and allows
the receiver to skip missing packets that have not been delivered in
time. The timeout of dropping a packet is based on the TSBPD
mechanism (see Section 4.5).
In the SRT, when Too-Late Packet Drop is enabled, and a packet
timestamp is older than 125% of the SRT latency, it is considered too
late to be delivered and may be dropped by the sender. However, the
sender keeps packets for at least 1 second in case the SRT latency is
not enough for a large RTT (that is, if 125% of the SRT latency is
less than 1 second).
When enabled on the receiver, the receiver drops packets that have
not been delivered or retransmitted in time, and delivers the
subsequent packets to the application when it is their time to play.
In pseudo-code, the algorithm of reading from the receiver buffer is
the following:
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<CODE BEGINS>
pos = 0; /* Current receiver buffer position */
i = 0; /* Position of the next available in the receiver buffer
packet relatively to the current buffer position pos */
while(True) {
// Get the position i of the next available packet
// in the receiver buffer
i = next_avail();
// Calculate packet delivery time PktTsbpdTime
// for the next available packet
PktTsbpdTime = delivery_time(i);
if T_NOW < PktTsbpdTime:
continue;
Drop packets which buffer position number is less than i;
Deliver packet with the buffer position i;
pos = i + 1;
}
<CODE ENDS>
where T_NOW is the current time according to the receiver clock.
The TLPKTDROP mechanism can be turned off to always ensure a clean
delivery. However, a lost packet can simply pause a delivery for
some longer, potentially undefined time, and cause even worse tearing
for the player. Setting higher SRT latency will help much more in
the case when TLPKTDROP causes packet drops too often.
4.7. Drift Management
When the sender enters "connected" status it tells the application
there is a socket interface that is transmitter-ready. At this point
the application can start sending data packets. It adds packets to
the SRT sender's buffer at a certain input rate, from which they are
transmitted to the receiver at scheduled times.
A synchronized time is required to keep proper sender/receiver buffer
levels, taking into account the time zone and round-trip time (up to
2 seconds for satellite links). Considering addition/subtraction
round-off, and possibly unsynchronized system times, an agreed-upon
time base drifts by a few microseconds every minute. The drift may
accumulate over many days to a point where the sender or receiver
buffers will overflow or deplete, seriously affecting the quality of
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the video. SRT has a time management mechanism to compensate for
this drift.
When a packet is received, SRT determines the difference between the
time it was expected and its timestamp. The timestamp is calculated
on the receiver side. The RTT tells the receiver how much time it
was supposed to take. SRT maintains a reference between the time at
the leading edge of the send buffer's latency window and the
corresponding time on the receiver (the present time). This allows
to convert packet timestamp to the local receiver time. Based on
this time, various events (packet delivery, etc.) can be scheduled.
The receiver samples time drift data and periodically calculates a
packet timestamp correction factor, which is applied to each data
packet received by adjusting the inter-packet interval. When a
packet is received it is not given right away to the application. As
time advances, the receiver knows the expected time for any missing
or dropped packet, and can use this information to fill any "holes"
in the receive queue with another packet (see Section 4.5).
It is worth noting that the period of sampling time drift data is
based on a number of packets rather than time duration to ensure
enough samples, independently of the media stream packet rate. The
effect of network jitter on the estimated time drift is attenuated by
using a large number of samples. The actual time drift being very
slow (affecting a stream only after many hours) does not require a
fast reaction.
The receiver uses local time to be able to schedule events -- to
determine, for example, if it is time to deliver a certain packet
right away. The timestamps in the packets themselves are just
references to the beginning of the session. When a packet is
received (with a timestamp from the sender), the receiver makes a
reference to the beginning of the session to recalculate its
timestamp. The start time is derived from the local time at the
moment that the session is connected. A packet timestamp equals
"now" minus "StartTime", where the latter is the point in time when
the socket was created.
4.8. Acknowledgement and Lost Packet Handling
To enable the Automatic Repeat reQuest of data packet
retransmissions, a sender stores all sent data packets in its buffer.
The SRT receiver periodically sends acknowledgements (ACKs) for the
received data packets so that the SRT sender can remove the
acknowledged packets from its buffer (Section 4.8.1). Once the
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acknowledged packets are removed, their retransmission is no longer
possible and presumably not needed.
Upon receiving the full acknowledgement (ACK) control packet, the SRT
sender should acknowledge its reception to the receiver by sending an
ACKACK control packet with the sequence number of the full ACK packet
being acknowledged.
The SRT receiver also sends NAK control packets to notify the sender
about the missing packets (Section 4.8.2). The sending of a NAK
packet can be triggered immediately after a gap in sequence numbers
of data packets is detected. In addition, a Periodic NAK report
mechanism can be used to send NAK reports periodically. The NAK
packet in that case will list all the packets that the receiver
considers being lost up to the moment the Periodic NAK report is
sent.
Upon reception of the NAK packet, the SRT sender prioritizes
retransmissions of lost packets over the regular data packets to be
transmitted for the first time.
The retransmission of the missing packet is repeated until the
receiver acknowledges its receipt, or if both peers agree to drop
this packet (see Section 4.6).
4.8.1. Packet Acknowledgement (ACKs, ACKACKs)
At certain intervals (see below), the SRT receiver sends an
acknowledgement (ACK) that causes the acknowledged packets to be
removed from the SRT sender's buffer.
An ACK control packet contains the sequence number of the packet
immediately following the latest in the list of received packets.
Where no packet loss has occurred up to the packet with sequence
number n, an ACK would include the sequence number (n + 1).
An ACK (from a receiver) will trigger the transmission of an ACKACK
(by the sender), with almost no delay. The time it takes for an ACK
to be sent and an ACKACK to be received is the RTT. The ACKACK tells
the receiver to stop sending the ACK position because the sender
already knows it. Otherwise, ACKs (with outdated information) would
continue to be sent regularly. Similarly, if the sender doesn't
receive an ACK, it doesn't stop transmitting.
There are two conditions for sending an acknowledgement. A full ACK
is based on a timer of 10 milliseconds (the ACK period). For high
bit rate transmissions, a "light ACK" can be sent, which is an ACK
for a sequence of packets. In a 10 milliseconds interval, there are
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often so many packets being sent and received that the ACK position
on the sender doesn't advance quickly enough. To mitigate this,
after 64 packets (even if the ACK period has not fully elapsed) the
receiver sends a light ACK. A light ACK is a shorter ACK (header + 1
x 32-bit field). It does not trigger an ACKACK.
When a receiver encounters the situation where the next packet to be
played was not successfully received from the sender, it will "skip"
this packet (see Section 4.6) and send a fake ACK. To the sender,
this fake ACK is a real ACK, and so it just behaves as if the packet
had been received. This facilitates the synchronization between SRT
sender and receiver. The fact that a packet was skipped remains
unknown by the sender. Skipped packets are recorded in the
statistics on the SRT receiver.
4.8.2. Packet Retransmission (NAKs)
The SRT receiver sends NAK control packets to notify the sender about
the missing packets. The NAK packet sending can be triggered
immediately after a gap in sequence numbers of data packets is
detected.
Upon reception of the NAK packet, the SRT sender prioritizes
retransmissions of lost packets over the regular data packets to be
transmitted for the first time.
The SRT sender maintains a list of lost packets (loss list) that is
built from NAK reports. When scheduling packet transmission, it
looks to see if a packet in the loss list has priority and sends it
if so. Otherwise, it sends the next packet from the scheduled for
the first transmission list. Note that when a packet is transmitted,
it stays in the buffer in case it is not received by the SRT
receiver.
NAK packets are processed to fill in the loss list. As the latency
window advances and packets are dropped from the sending queue, a
check is performed to see if any of the dropped or resent packets are
in the loss list, to determine if they can be removed from there as
well so that they are not retransmitted unnecessarily.
There is a counter for the packets that are resent. If there is no
ACK for a packet, it will stay in the loss list and can be resent
more than once. Packets in the loss list are prioritized.
If packets in the loss list continue to block the send queue, at some
point this will cause the send queue to fill. When the send queue is
full, the sender will begin to drop packets without even sending them
the first time. An encoder (or other application) may continue to
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provide packets, but there's no place for them, so they will end up
being thrown away.
This condition where packets are unsent doesn't happen often. There
is a maximum number of packets held in the send buffer based on the
configured latency. Older packets that have no chance to be
retransmitted and played in time are dropped, making room for newer
real-time packets produced by the sending application. See sections
Section 4.5, Section 4.6 for details.
In addition to the regular NAKs, the Periodic NAK report mechanism
can be used to send NAK reports periodically. The NAK packet in that
case will have all the packets that the receiver considers being lost
at the time of sending the Periodic NAK report.
An ACKACK tells the receiver to stop sending the ACK position because
the sender already knows it. Otherwise, ACKs (with outdated
information) would continue to be sent regularly.
An ACK serves as a ping, with a corresponding ACKACK pong, to measure
RTT. The time it takes for an ACK to be sent and an ACKACK to be
received is the RTT. Each ACK has a number. A corresponding ACKACK
has that same number. The receiver keeps a list of all ACKs in a
queue to match them. Unlike a full ACK, which contains the current
RTT and several other values in the CIF, a light ACK just contains
the sequence number. All control messages are sent directly and
processed upon reception, but ACKACK processing time is negligible
(the time this takes is included in the round-trip time).
4.9. Bidirectional Transmission Queues
Once an SRT connection is established, both peers can send data
packets simultaneously.
4.10. Round Trip Time Estimation
The round-trip time is estimated during the transmission of SRT data
packets based on the time difference between the ACK packet is sent
and the corresponding ACKACK is received by the data receiver.
4.11. Congestion Control
SRT provides certain mechanisms for the sender to get some feedback
from the receiving side through the ACK packets (Section 3.2.3).
Every 10 ms the sender receives the latest values of RTT and RTT
variance, Available Buffer Size, Packets Receiving Rate and Estimated
Link Capacity. Upon reception of the NAK packet (Section 3.2.4) the
sender can detect packet losses during the transmission. These
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mechanisms provide a solid background for various congestion control
algorithms.
Given that SRT can operate in live and file transfer modes, there are
two groups of congestion control algorithms possible.
For live transmission mode (Section 4.2.2) the congestion control
algorithm does not need to control the sending pace of the data
packets, as the sending timing is provided by the live input.
Although certain limitations on the minimal inter-sending time of
consecutive packets can be applied in order to avoid congestion
during fluctuations of the source bitrate. Also it is allowed to
drop those packets that can not be delivered in time.
For file transfer, any known File Congestion Control algorithms like
CUBIC and BBR can apply, including the congestion control mechanism
proposed in UDT [GHG04b]. The UDT congestion control relies on the
available link capacity, packet loss reports (NAK) and packet
acknowledgements (ACKs). It then slows down the output of packets as
needed by adjusting the packet sending pace. In periods of
congestion, it can block the main stream and focus on the lost
packets.
5. Encryption
SRT supports encryption based on a pre-shared secret. Please refer
to [SRTTO] for more information.
6. Security Considerations
SRT supports confidentiality of user data using stream ciphering
based on AES. Session keys for ciphering are delivered through
control packets during handshake, with the protection by Key
Encryption Key, which is generated by a sender and receiver with pre-
shared secret such as passphrase. As in UDT, careful uses of SYN
Cookies may help to deter denial of service attacks. Appropriate
security policy including key size, key refresh period, as well as
passphrase should be managed by security officers, which is out of
scope of the present document.
7. IANA Considerations
This document makes no requests of the IANA.
Contributors
This specification is heavily based on the SRT Protocol Technical
Overview [SRTTO] written by Jean Dube and Steve Matthews.
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In alphabetical order, the contributors to the pre-IETF SRT project
and specification at Haivision are: Marc Cymontkowski, Roman
Diouskine, Jean Dube, Mikolaj Malecki, Steve Matthews, Maria
Sharabayko, Maxim Sharabayko, Adam Yellen.
The contributors to this specification at SK Telecom are Jeongseok
Kim and Joonwoong Kim.
We cannot list all the contributors to the open-sourced
implementation of SRT on GitHub. But we appreciate the help,
contribution, integrations and feedback of the SRT and SRT Alliances
community.
Acknowledgments
The basis of the SRT protocol and its implementation was the UDP-
based Data Transfer Protocol [GHG04b]. The authors thank Yunhong Gu
and Robert Grossman, the authors of the UDP-based Data Transfer
Protocol [GHG04b].
TODO acknowledge.
References
Normative References
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980,
<https://www.rfc-editor.org/info/rfc768>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
Informative References
[AV1] Rivaz, P.d. and J. Haughton, "AV1 Bitstream & Decoding
Process Specification", March 2020,
<https://aomediacodec.github.io/av1-spec/av1-spec.pdf>.
[GHG04b] Gu, Y., Hong, X., and R.L. Grossman,, "Experiences in
Design and Implementation of a High Performance Transport
Protocol", DOI 10.1109/SC.2004.24, December 2004,
<https://doi.org/10.1109/SC.2004.24>.
[H.265] International Telecommunications Union, "H.265 : High
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efficiency video coding", ITU-T Recommendation H.265,
2019.
[I-D.ietf-quic-http]
Bishop, M., "Hypertext Transfer Protocol Version 3
(HTTP/3)", Work in Progress, Internet-Draft, draft-ietf-
quic-http-27, 21 February 2020, <http://www.ietf.org/
internet-drafts/draft-ietf-quic-http-27.txt>.
[I-D.ietf-quic-transport]
Iyengar, J. and M. Thomson, "QUIC: A UDP-Based Multiplexed
and Secure Transport", Work in Progress, Internet-Draft,
draft-ietf-quic-transport-27, 21 February 2020,
<http://www.ietf.org/internet-drafts/draft-ietf-quic-
transport-27.txt>.
[ISO13818-1]
ISO, "Information technology -- Generic coding of moving
pictures and associated audio information: Systems", ISO/
IEC 13818-1, March 2020.
[ISO23009] ISO, "Information technology -- Dynamic adaptive streaming
over HTTP (DASH)", ISO/IEC 23009:2019, March 2020.
[PNPID] "PNP ID AND ACPI ID REGISTRY", March 2020,
<https://uefi.org/PNP_ACPI_Registry>.
[RFC3031] Rosen, E., Viswanathan, A., and R. Callon, "Multiprotocol
Label Switching Architecture", RFC 3031,
DOI 10.17487/RFC3031, January 2001,
<https://www.rfc-editor.org/info/rfc3031>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC8216] Pantos, R., Ed. and W. May, "HTTP Live Streaming",
RFC 8216, DOI 10.17487/RFC8216, August 2017,
<https://www.rfc-editor.org/info/rfc8216>.
[RTMP] "Real-Time Messaging Protocol", March 2020,
<https://www.adobe.com/devnet/rtmp.html>.
[SP800-38A]
Dworkin, M., "Recommendation for Block Cipher Modes of
Operation", December 2001.
Sharabayko, et al. Expires 10 September 2020 [Page 48]
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[SRTSRC] "SRT fully functional reference implementation", March
2020, <https://github.com/Haivision/srt>.
[SRTTO] Dube, J. and S. Matthews, "SRT Protocol Technical
Overview", December 2019.
[VP9] WebM, "VP9 Video Codec", March 2020,
<https://www.webmproject.org/vp9>.
Appendix A. Packet Sequence List coding
For any single packet sequence number, it uses the original sequence
number in the field. The first bit must start with "0".
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Sequence Number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 15: single sequence numbers coding
For any consecutive packet sequence numbers that the difference
between the last and first is more than 1, only record the first (a)
and the the last (b) sequence numbers in the list field, and modify
the the first bit of a to "1".
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| Sequence Number a (first) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| Sequence Number b (last) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 16: list of sequence numbers coding
Authors' Addresses
Maxim Sharabayko
Haivision Network Video, GmbH
Email: maxsharabayko@haivision.com
Maria Sharabayko
Haivision Network Video, GmbH
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Email: msharabayko@haivision.com
Jean Dube
Haivision
Email: jdube@haivision.com
Jeongseok Kim
SK Telecom Co., Ltd.
Email: jeongseok.kim@sk.com
Joonwoong Kim
SK Telecom Co., Ltd.
Email: joonwoong.kim@sk.com
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