| Rfc | 7501 |
|---|
| Title | Terminology for Benchmarking Session Initiation Protocol (SIP)
Devices: Basic Session Setup and Registration |
|---|
| Author | C. Davids, V.
Gurbani, S. Poretsky |
|---|
| Date | April 2015 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | INFORMATIONAL |
|---|
|
Internet Engineering Task Force (IETF) C. Davids
Request for Comments: 7501 Illinois Institute of Technology
Category: Informational V. Gurbani
ISSN: 2070-1721 Bell Laboratories, Alcatel-Lucent
S. Poretsky
Allot Communications
April 2015
Terminology for Benchmarking Session Initiation Protocol (SIP) Devices:
Basic Session Setup and Registration
Abstract
This document provides a terminology for benchmarking the Session
Initiation Protocol (SIP) performance of devices. Methodology
related to benchmarking SIP devices is described in the companion
methodology document (RFC 7502). Using these two documents,
benchmarks can be obtained and compared for different types of
devices such as SIP Proxy Servers, Registrars, and Session Border
Controllers. The term "performance" in this context means the
capacity of the Device Under Test (DUT) to process SIP messages.
Media streams are used only to study how they impact the signaling
behavior. The intent of the two documents is to provide a normalized
set of tests that will enable an objective comparison of the capacity
of SIP devices. Test setup parameters and a methodology are
necessary because SIP allows a wide range of configurations and
operational conditions that can influence performance benchmark
measurements. A standard terminology and methodology will ensure
that benchmarks have consistent definitions and were obtained
following the same procedures.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7501.
Copyright Notice
Copyright (c) 2015 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
(http://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 and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Term Definitions . . . . . . . . . . . . . . . . . . . . . . 7
3.1. Protocol Components . . . . . . . . . . . . . . . . . . . 7
3.1.1. Session . . . . . . . . . . . . . . . . . . . . . . . 7
3.1.2. Signaling Plane . . . . . . . . . . . . . . . . . . . 8
3.1.3. Media Plane . . . . . . . . . . . . . . . . . . . . . 8
3.1.4. Associated Media . . . . . . . . . . . . . . . . . . 9
3.1.5. Overload . . . . . . . . . . . . . . . . . . . . . . 9
3.1.6. Session Attempt . . . . . . . . . . . . . . . . . . . 10
3.1.7. Established Session . . . . . . . . . . . . . . . . . 10
3.1.8. Session Attempt Failure . . . . . . . . . . . . . . . 11
3.2. Test Components . . . . . . . . . . . . . . . . . . . . . 11
3.2.1. Emulated Agent . . . . . . . . . . . . . . . . . . . 11
3.2.2. Signaling Server . . . . . . . . . . . . . . . . . . 12
3.2.3. SIP Transport Protocol . . . . . . . . . . . . . . . 12
3.3. Test Setup Parameters . . . . . . . . . . . . . . . . . . 13
3.3.1. Session Attempt Rate . . . . . . . . . . . . . . . . 13
3.3.2. Establishment Threshold Time . . . . . . . . . . . . 13
3.3.3. Session Duration . . . . . . . . . . . . . . . . . . 14
3.3.4. Media Packet Size . . . . . . . . . . . . . . . . . . 14
3.3.5. Codec Type . . . . . . . . . . . . . . . . . . . . . 15
3.4. Benchmarks . . . . . . . . . . . . . . . . . . . . . . . 15
3.4.1. Session Establishment Rate . . . . . . . . . . . . . 15
3.4.2. Registration Rate . . . . . . . . . . . . . . . . . . 16
3.4.3. Registration Attempt Rate . . . . . . . . . . . . . . 17
4. Security Considerations . . . . . . . . . . . . . . . . . . . 17
5. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
5.1. Normative References . . . . . . . . . . . . . . . . . . 18
5.2. Informative References . . . . . . . . . . . . . . . . . 18
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 20
1. Introduction
Service Providers and IT organizations deliver Voice Over IP (VoIP)
and multimedia network services based on the IETF Session Initiation
Protocol (SIP) [RFC3261]. SIP is a signaling protocol originally
intended to be used to dynamically establish, disconnect, and modify
streams of media between end users. As it has evolved, it has been
adopted for use in a growing number of services and applications.
Many of these result in the creation of a media session, but some do
not. Examples of this latter group include text messaging and
subscription services. The set of benchmarking terms provided in
this document is intended for use with any SIP-enabled device
performing SIP functions in the interior of the network, whether or
not these result in the creation of media sessions. The performance
of end-user devices is outside the scope of this document.
A number of networking devices have been developed to support SIP-
based VoIP services. These include SIP servers, Session Border
Controllers (SBCs), and Back-to-back User Agents (B2BUAs). These
devices contain a mix of voice and IP functions whose performance may
be reported using metrics defined by the equipment manufacturer or
vendor. The Service Provider or IT organization seeking to compare
the performance of such devices will not be able to do so using these
vendor-specific metrics, whose conditions of test and algorithms for
collection are often unspecified.
SIP functional elements and the devices that include them can be
configured many different ways and can be organized into various
topologies. These configuration and topological choices impact the
value of any chosen signaling benchmark. Unless these conditions of
test are defined, a true comparison of performance metrics across
multiple vendor implementations will not be possible.
Some SIP-enabled devices terminate or relay media as well as
signaling. The processing of media by the device impacts the
signaling performance. As a result, the conditions of test must
include information as to whether or not the Device Under Test
processes media. If the device processes media during the test, a
description of the media must be provided. This document and its
companion methodology document [RFC7502] provide a set of black-box
benchmarks for describing and comparing the performance of devices
that incorporate the SIP User Agent Client and Server functions and
that operate in the network's core.
The definition of SIP performance benchmarks necessarily includes
definitions of Test Setup Parameters and a test methodology. These
enable the Tester to perform benchmarking tests on different devices
and to achieve comparable results. This document provides a common
set of definitions for Test Components, Test Setup Parameters, and
Benchmarks. All the benchmarks defined are black-box measurements of
the SIP signaling plane. The Test Setup Parameters and Benchmarks
defined in this document are intended for use with the companion
methodology document.
1.1. Scope
The scope of this document is summarized as follows:
o This terminology document describes SIP signaling performance
benchmarks for black-box measurements of SIP networking devices.
Stress conditions and debugging scenarios are not addressed in
this document.
o The DUT must be network equipment that is RFC 3261 capable. This
may be a Registrar, Redirect Server, or Stateful Proxy. This
document does not require the intermediary to assume the role of a
stateless proxy. A DUT may also act as a B2BUA or take the role
of an SBC.
o The Tester acts as multiple Emulated Agents (EAs) that initiate
(or respond to) SIP messages as session endpoints and source (or
receive) associated media for established connections.
o Regarding SIP signaling in presence of media:
* The media performance is not benchmarked.
* Some tests require media, but the use of media is limited to
observing the performance of SIP signaling. Tests that require
media will annotate the media characteristics as a condition of
test.
* The type of DUT dictates whether the associated media streams
traverse the DUT. Both scenarios are within the scope of this
document.
* SIP is frequently used to create media streams; the signaling
plane and media plane are treated as orthogonal to each other
in this document. While many devices support the creation of
media streams, benchmarks that measure the performance of these
streams are outside the scope of this document and its
companion methodology document [RFC7502]. Tests may be
performed with or without the creation of media streams. The
presence or absence of media streams MUST be noted as a
condition of the test, as the performance of SIP devices may
vary accordingly. Even if the media is used during
benchmarking, only the SIP performance will be benchmarked, not
the media performance or quality.
o Both INVITE and non-INVITE scenarios (registrations) are addressed
in this document. However, benchmarking SIP presence or
subscribe-notify extensions is not a part of this document.
o Different transport -- such as UDP, TCP, SCTP, or TLS -- may be
used. The specific transport mechanism MUST be noted as a
condition of the test, as the performance of SIP devices may vary
accordingly.
o REGISTER and INVITE requests may be challenged or remain
unchallenged for authentication purposes. Whether or not the
REGISTER and INVITE requests are challenged is a condition of test
that will be recorded along with other such parameters that may
impact the SIP performance of the device or system under test.
o Re-INVITE requests are not considered within the scope of this
document since the benchmarks for INVITEs are based on the dialog
created by the INVITE and not on the transactions that take place
within that dialog.
o Only session establishment is considered for the performance
benchmarks. Session disconnect is not considered within the scope
of this document. This is because our goal is to determine the
maximum capacity of the device or system under test, that is, the
number of simultaneous SIP sessions that the device or system can
support. It is true that there are BYE requests being created
during the test process. These transactions do contribute to the
load on the device or system under test and thus are accounted for
in the metric we derive. We do not seek a separate metric for the
number of BYE transactions a device or system can support.
o Scenarios that are specific to the IP Multimedia Subsystem (IMS)
are not considered, but test cases can be applied with 3GPP-
specific SIP signaling and the Proxy-Call Session Control Function
(P-CSCF) as a DUT.
o The benchmarks described in this document are intended for a
laboratory environment and are not intended to be used on a
production network. Some of the benchmarks send enough traffic
that a denial-of-service attack is possible if used in production
networks.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14, RFC2119
[RFC2119]. RFC 2119 defines the use of these key words to help make
the intent of Standards Track documents as clear as possible. While
this document uses these keywords, this document is not a Standards
Track document.
For the sake of clarity and continuity, this document adopts the
template for definitions set out in Section 2 of RFC 1242 [RFC1242].
The term "Device Under Test (DUT)" is defined in Section 3.1.1 of RFC
2285 [RFC2285].
Many commonly used SIP terms in this document are defined in RFC 3261
[RFC3261]. For convenience, the most important of these are
reproduced below. Use of these terms in this document is consistent
with their corresponding definition in the base SIP specification
[RFC3261] as amended by [RFC4320], [RFC5393], and [RFC6026].
o Call Stateful: A proxy is call stateful if it retains state for a
dialog from the initiating INVITE to the terminating BYE request.
A call stateful proxy is always transaction stateful, but the
converse is not necessarily true.
o Stateful Proxy: A logical entity, as defined by [RFC3261], that
maintains the client and server transaction state machines during
the processing of a request. (Also known as a transaction
stateful proxy.) The behavior of a stateful proxy is further
defined in Section 16 of RFC 3261 [RFC3261] . A transaction
stateful proxy is not the same as a call stateful proxy.
o Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
logical entity that receives a request and processes it as a user
agent server (UAS). In order to determine how the request should
be answered, it acts as a user agent client (UAC) and generates
requests. Unlike a proxy server, it maintains dialog state and
must participate in all requests sent on the dialogs it has
established. Since it is a concatenation of a UAC and a UAS, no
explicit definitions are needed for its behavior.
3. Term Definitions
3.1. Protocol Components
3.1.1. Session
Definition:
The combination of signaling and media messages and associated
processing that enable a single SIP-based audio or video call, or
SIP registration.
Discussion:
The term "session" commonly implies a media session. In this
document the term is extended to cover the signaling and any media
specified and invoked by the corresponding signaling.
Measurement Units:
N/A.
Issues:
None.
See Also:
Media Plane
Signaling Plane
Associated Media
3.1.2. Signaling Plane
Definition:
The plane in which SIP messages [RFC3261] are exchanged between
SIP agents [RFC3261].
Discussion:
SIP messages are used to establish sessions in several ways:
directly between two User Agents [RFC3261], through a Proxy Server
[RFC3261], or through a series of Proxy Servers. The Session
Description Protocol (SDP) is included in the Signaling Plane.
Measurement Units:
N/A.
Issues:
None.
See Also:
Media Plane
Emulated Agent
3.1.3. Media Plane
Definition:
The data plane in which one or more media streams and their
associated media control protocols (e.g., RTCP [RFC3550]) are
exchanged between User Agents after a media connection has been
created by the exchange of signaling messages in the Signaling
Plane.
Discussion:
Media may also be known as the "bearer channel". The Media Plane
MUST include the media control protocol, if one is used, and the
media stream(s). Examples of media are audio and video. The
media streams are described in the SDP of the Signaling Plane.
Measurement Units:
N/A.
Issues:
None.
See Also:
Signaling Plane
3.1.4. Associated Media
Definition:
Media that corresponds to an 'm' line in the SDP payload of the
Signaling Plane.
Discussion:
The format of the media is determined by the SDP attributes for
the corresponding 'm' line.
Measurement Units:
N/A.
Issues:
None.
3.1.5. Overload
Definition:
Overload is defined as the state where a SIP server does not have
sufficient resources to process all incoming SIP messages
[RFC6357].
Discussion:
The distinction between an overload condition and other failure
scenarios is outside the scope of black-box testing and of this
document. Under overload conditions, all or a percentage of
Session Attempts will fail due to lack of resources. In black-box
testing, the cause of the failure is not explored. The fact that
a failure occurred for whatever reason will trigger the tester to
reduce the offered load, as described in the companion methodology
document [RFC7502]. SIP server resources may include CPU
processing capacity, network bandwidth, input/output queues, or
disk resources. Any combination of resources may be fully
utilized when a SIP server (the DUT) is in the overload condition.
For proxy-only (or intermediary) devices, it is expected that the
proxy will be driven into overload based on the delivery rate of
signaling requests.
Measurement Units:
N/A.
3.1.6. Session Attempt
Definition:
A SIP INVITE or REGISTER request sent by the EA that has not
received a final response.
Discussion:
The attempted session may be either an invitation to an audio/
video communication or a registration attempt. When counting the
number of session attempts, we include all requests that are
rejected for lack of authentication information. The EA needs to
record the total number of session attempts including those
attempts that are routinely rejected by a proxy that requires the
UA to authenticate itself. The EA is provisioned to deliver a
specific number of session attempts per second. But the EA must
also count the actual number of session attempts per given time
interval.
Measurement Units:
N/A.
Issues:
None.
See Also:
Session
Session Attempt Rate
3.1.7. Established Session
Definition:
A SIP session for which the EA acting as the UA has received a 200
OK message.
Discussion:
An Established Session may be either an invitation to an audio/
video communication or a registration attempt. Early dialogs for
INVITE requests are out of scope for this work.
Measurement Units:
N/A.
Issues:
None.
See Also:
None.
3.1.8. Session Attempt Failure
Definition:
A session attempt that does not result in an Established Session.
Discussion:
The session attempt failure may be indicated by the following
observations at the EA:
1. Receipt of a SIP 3xx-, 4xx-, 5xx-, or 6xx-class response to a
Session Attempt.
2. The lack of any received SIP response to a Session Attempt
within the Establishment Threshold Time (cf. Section 3.3.2).
Measurement Units:
N/A.
Issues:
None.
See Also:
Session Attempt
3.2. Test Components
3.2.1. Emulated Agent
Definition:
A device in the test topology that initiates/responds to SIP
messages as one or more session endpoints and, wherever
applicable, sources/receives Associated Media for Established
Sessions.
Discussion:
The EA functions in the Signaling and Media Planes. The Tester
may act as multiple EAs.
Measurement Units:
N/A.
Issues:
None.
See Also:
Media Plane
Signaling Plane
Established Session
Associated Media
3.2.2. Signaling Server
Definition:
Device in the test topology that facilitates the creation of
sessions between EAs. This device is the DUT.
Discussion:
The DUT is a network intermediary that is RFC 3261 capable such as
a Registrar, Redirect Server, Stateful Proxy, B2BUA, or SBC.
Measurement Units:
N/A.
Issues:
None.
See Also:
Signaling Plane
3.2.3. SIP Transport Protocol
Definition:
The protocol used for transport of the Signaling Plane messages.
Discussion:
Performance benchmarks may vary for the same SIP networking device
depending upon whether TCP, UDP, TLS, SCTP, websockets [RFC7118],
or any future transport-layer protocol is used. For this reason,
it is necessary to measure the SIP Performance Benchmarks using
these various transport protocols. Performance Benchmarks MUST
report the SIP Transport Protocol used to obtain the benchmark
results.
Measurement Units:
While these are not units of measure, they are attributes that are
one of many factors that will contribute to the value of the
measurements to be taken. TCP, UDP, SCTP, TLS over TCP, TLS over
UDP, TLS over SCTP, and websockets are among the possible values
to be recorded as part of the test.
Issues:
None.
See Also:
None.
3.3. Test Setup Parameters
3.3.1. Session Attempt Rate
Definition:
Configuration of the EA for the number of sessions per second
(sps) that the EA attempts to establish using the services of the
DUT.
Discussion:
The Session Attempt Rate is the number of sessions per second that
the EA sends toward the DUT. Some of the sessions attempted may
not result in a session being established.
Measurement Units:
Session Attempts per second
Issues:
None.
See Also:
Session
Session Attempt
3.3.2. Establishment Threshold Time
Definition:
Configuration of the EA that represents the amount of time that an
EA client will wait for a response from an EA server before
declaring a Session Attempt Failure.
Discussion:
This time duration is test dependent.
It is RECOMMENDED that the Establishment Threshold Time value be
set to Timer B or Timer F as specified in RFC 3261, Table 4
[RFC3261].
Measurement Units:
seconds
Issues:
None.
See Also:
None.
3.3.3. Session Duration
Definition:
Configuration of the EA that represents the amount of time that
the SIP dialog is intended to exist between the two EAs associated
with the test.
Discussion:
The time at which the BYE is sent will control the Session
Duration.
Measurement Units:
seconds
Issues:
None.
See Also:
None.
3.3.4. Media Packet Size
Definition:
Configuration on the EA for a fixed number of frames or samples to
be sent in each RTP packet of the media stream when the test
involves Associated Media.
Discussion:
This document describes a method to measure SIP performance. If
the DUT is processing media as well as SIP messages the media
processing will potentially slow down the SIP processing and lower
the SIP performance metric. The tests with associated media are
designed for audio codecs, and the assumption was made that larger
media packets would require more processor time. This document
does not define parameters applicable to video codecs.
For a single benchmark test, media sessions use a defined number
of samples or frames per RTP packet. If two SBCs, for example,
used the same codec but one puts more frames into the RTP packet,
this might cause variation in the performance benchmark results.
Measurement Units:
An integer number of frames or samples, depending on whether a
hybrid- or sample-based codec is used, respectively.
Issues:
None.
See Also:
None.
3.3.5. Codec Type
Definition:
The name of the codec used to generate the media session.
Discussion:
For a single benchmark test, all sessions use the same size packet
for media streams. The size of packets can cause a variation in
the performance benchmark measurements.
Measurement Units:
This is a textual name (alphanumeric) assigned to uniquely
identify the codec.
Issues:
None.
See Also:
None.
3.4. Benchmarks
3.4.1. Session Establishment Rate
Definition:
The maximum value of the Session Attempt Rate that the DUT can
handle for an extended, predefined period with zero failures.
Discussion:
This benchmark is obtained with zero failure. The Session Attempt
Rate provisioned on the EA is raised and lowered as described in
the algorithm in the accompanying methodology document [RFC7502],
until a traffic load over the period of time necessary to attempt
N sessions completes without failure, where N is a parameter
specified in the algorithm and recorded in the Test Setup Report.
Measurement Units:
sessions per second (sps)
Issues:
None.
See Also:
Session Attempt Rate
3.4.2. Registration Rate
Definition:
The maximum value of the Registration Attempt Rate that the DUT
can handle for an extended, predefined period with zero failures.
Discussion:
This benchmark is obtained with zero failures. The registration
rate provisioned on the Emulated Agent is raised and lowered as
described in the algorithm in the companion methodology document
[RFC7502], until a traffic load consisting of registration
attempts at the given attempt rate over the period of time
necessary to attempt N registrations completes without failure,
where N is a parameter specified in the algorithm and recorded in
the Test Setup Report.
This benchmark is described separately from the Session
Establishment Rate (Section 3.4.1), although it could be
considered a special case of that benchmark, since a REGISTER
request is a request for a session that is not initiated by an
INVITE request. It is defined separately because it is a very
important benchmark for most SIP installations. An example
demonstrating its use is an avalanche restart, where hundreds of
thousands of endpoints register simultaneously following a power
outage. In such a case, an authoritative measurement of the
capacity of the device to register endpoints is useful to the
network designer. Additionally, in certain controlled networks,
there appears to be a difference between the registration rate of
new endpoints and the registering rate of existing endpoints
(register refreshes). This benchmark can capture these
differences as well.
Measurement Units:
registrations per second (rps)
Issues:
None.
See Also:
None.
3.4.3. Registration Attempt Rate
Definition:
Configuration of the EA for the number of registrations per second
that the EA attempts to send to the DUT.
Discussion:
The Registration Attempt Rate is the number of registration
requests per second that the EA sends toward the DUT.
Measurement Units:
registrations per second (rps)
Issues:
None.
See Also:
None.
4. Security Considerations
Documents of this type do not directly affect the security of the
Internet or corporate networks as long as benchmarking is not
performed on devices or systems connected to production networks.
Security threats and how to counter these in SIP and the media layer
are discussed in RFC 3261 [RFC3261], RFC 3550 [RFC3550], and RFC 3711
[RFC3711]. This document attempts to formalize a set of common
terminology for benchmarking SIP networks. Packets with unintended
and/or unauthorized DSCP or IP precedence values may present security
issues. Determining the security consequences of such packets is out
of scope for this document.
5. References
5.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002, <http://www.rfc-editor.org/info/rfc3261>.
[RFC5393] Sparks, R., Ed., Lawrence, S., Hawrylyshen, A., and B.
Campen, "Addressing an Amplification Vulnerability in
Session Initiation Protocol (SIP) Forking Proxies", RFC
5393, December 2008,
<http://www.rfc-editor.org/info/rfc5393>.
[RFC4320] Sparks, R., "Actions Addressing Identified Issues with the
Session Initiation Protocol's (SIP) Non-INVITE
Transaction", RFC 4320, January 2006,
<http://www.rfc-editor.org/info/rfc4320>.
[RFC6026] Sparks, R. and T. Zourzouvillys, "Correct Transaction
Handling for 2xx Responses to Session Initiation Protocol
(SIP) INVITE Requests", RFC 6026, September 2010,
<http://www.rfc-editor.org/info/rfc6026>.
[RFC7502] Davids, C., Gurbani, V., and S. Poretsky, "Terminology for
Benchmarking Session Initiation Protocol (SIP) Devices:
Basic Session Setup and Registration", RFC 7502, April
2015, <http://www.rfc-editor.org/info/rfc7502>.
5.2. Informative References
[RFC2285] Mandeville, R., "Benchmarking Terminology for LAN
Switching Devices", RFC 2285, February 1998,
<http://www.rfc-editor.org/info/rfc2285>.
[RFC1242] Bradner, S., "Benchmarking Terminology for Network
Interconnection Devices", RFC 1242, July 1991,
<http://www.rfc-editor.org/info/rfc1242>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003,
<http://www.rfc-editor.org/info/rfc3550>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC6357] Hilt, V., Noel, E., Shen, C., and A. Abdelal, "Design
Considerations for Session Initiation Protocol (SIP)
Overload Control", RFC 6357, August 2011,
<http://www.rfc-editor.org/info/rfc6357>.
[RFC7118] Baz Castillo, I., Millan Villegas, J., and V. Pascual,
"The WebSocket Protocol as a Transport for the Session
Initiation Protocol (SIP)", RFC 7118, January 2014,
<http://www.rfc-editor.org/info/rfc7118>.
Acknowledgments
The authors would like to thank Keith Drage, Cullen Jennings, Daryl
Malas, Al Morton, and Henning Schulzrinne for invaluable
contributions to this document. Dale Worley provided an extensive
review that lead to improvements in the documents. We are grateful
to Barry Constantine, William Cerveny, and Robert Sparks for
providing valuable comments during the documents' last calls and
expert reviews. Al Morton and Sarah Banks have been exemplary
working group chairs; we thank them for tracking this work to
completion.
Authors' Addresses
Carol Davids
Illinois Institute of Technology
201 East Loop Road
Wheaton, IL 60187
United States
Phone: +1 630 682 6024
EMail: davids@iit.edu
Vijay K. Gurbani
Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane
Rm 9C-533
Naperville, IL 60566
United States
Phone: +1 630 224 0216
EMail: vkg@bell-labs.com
Scott Poretsky
Allot Communications
300 TradeCenter, Suite 4680
Woburn, MA 08101
United States
Phone: +1 508 309 2179
EMail: sporetsky@allot.com