< draft-ietf-sipping-rtcp-summary-12.txt   draft-ietf-sipping-rtcp-summary-13.txt >
SIPPING WG A. Pendleton SIPPING WG A. Pendleton
Internet-Draft A. Clark Internet-Draft A. Clark
Intended status: Standards Track Telchemy Incorporated Intended status: Standards Track Telchemy Incorporated
Expires: January 3, 2011 A. Johnston Expires: February 5, 2011 A. Johnston
Avaya Avaya
H. Sinnreich H. Sinnreich
Unaffiliated Unaffiliated
July 2, 2010 August 4, 2010
Session Initiation Protocol Event Package for Voice Quality Reporting Session Initiation Protocol Event Package for Voice Quality Reporting
draft-ietf-sipping-rtcp-summary-12 draft-ietf-sipping-rtcp-summary-13
Abstract Abstract
This document defines a Session Initiation Protocol (SIP) event This document defines a Session Initiation Protocol (SIP) event
package that enables the collection and reporting of metrics that package that enables the collection and reporting of metrics that
measure the quality for Voice over Internet Protocol (VoIP) sessions. measure the quality for Voice over Internet Protocol (VoIP) sessions.
Voice call quality information derived from RTP Control Protocol Voice call quality information derived from RTP Control Protocol
Extended Reports (RTCP-XR) and call information from SIP is conveyed Extended Reports (RTCP-XR) and call information from SIP is conveyed
from a User Agent (UA) in a session, known as a reporter, to a third from a User Agent (UA) in a session, known as a reporter, to a third
party, known as a collector. A registration for the application/ party, known as a collector. A registration for the application/
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Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 3, 2011. This Internet-Draft will expire on February 5, 2011.
Copyright Notice Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the Copyright (c) 2010 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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3.2. PUBLISH Method . . . . . . . . . . . . . . . . . . . . . . 6 3.2. PUBLISH Method . . . . . . . . . . . . . . . . . . . . . . 6
3.3. Multi-Party and Multi-Segment Calls . . . . . . . . . . . 7 3.3. Multi-Party and Multi-Segment Calls . . . . . . . . . . . 7
3.4. Overload Avoidance . . . . . . . . . . . . . . . . . . . . 7 3.4. Overload Avoidance . . . . . . . . . . . . . . . . . . . . 7
4. Event Package Formal Definition . . . . . . . . . . . . . . . 8 4. Event Package Formal Definition . . . . . . . . . . . . . . . 8
4.1. Event Package Name . . . . . . . . . . . . . . . . . . . . 8 4.1. Event Package Name . . . . . . . . . . . . . . . . . . . . 8
4.2. Event Package Parameters . . . . . . . . . . . . . . . . . 8 4.2. Event Package Parameters . . . . . . . . . . . . . . . . . 8
4.3. SUBSCRIBE Bodies . . . . . . . . . . . . . . . . . . . . . 8 4.3. SUBSCRIBE Bodies . . . . . . . . . . . . . . . . . . . . . 8
4.4. Subscribe Duration . . . . . . . . . . . . . . . . . . . . 8 4.4. Subscribe Duration . . . . . . . . . . . . . . . . . . . . 8
4.5. NOTIFY Bodies . . . . . . . . . . . . . . . . . . . . . . 8 4.5. NOTIFY Bodies . . . . . . . . . . . . . . . . . . . . . . 8
4.6. Voice Quality Event and Semantics . . . . . . . . . . . . 9 4.6. Voice Quality Event and Semantics . . . . . . . . . . . . 9
4.6.1. ABNF Syntax Definition . . . . . . . . . . . . . . . . 9 4.6.1. ABNF Syntax Definition . . . . . . . . . . . . . . . . 10
4.6.2. Parameter Definitions and Mappings . . . . . . . . . . 20 4.6.2. Parameter Definitions and Mappings . . . . . . . . . . 20
4.7. Message Flow and Syntax Examples . . . . . . . . . . . . . 28 4.7. Message Flow and Syntax Examples . . . . . . . . . . . . . 28
4.7.1. End of Session Report using NOTIFY . . . . . . . . . . 28 4.7.1. End of Session Report using NOTIFY . . . . . . . . . . 29
4.7.2. Mid Session Threshold Violation using NOTIFY . . . . . 30 4.7.2. Mid Session Threshold Violation using NOTIFY . . . . . 31
4.7.3. End of Session Report using PUBLISH . . . . . . . . . 33 4.7.3. End of Session Report using PUBLISH . . . . . . . . . 33
4.7.4. Alert Report using PUBLISH . . . . . . . . . . . . . . 35 4.7.4. Alert Report using PUBLISH . . . . . . . . . . . . . . 35
4.8. Configuration Dataset for vq-rtcpxr Events . . . . . . . . 37 4.8. Configuration Dataset for vq-rtcpxr Events . . . . . . . . 37
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37
5.1. SIP Event Package Registration . . . . . . . . . . . . . . 37 5.1. SIP Event Package Registration . . . . . . . . . . . . . . 37
5.2. application/vq-rtcp-xr MIME Registration . . . . . . . . . 38 5.2. application/vq-rtcp-xr MIME Registration . . . . . . . . . 38
6. Security Considerations . . . . . . . . . . . . . . . . . . . 38 6. Security Considerations . . . . . . . . . . . . . . . . . . . 38
7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 38 7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 38
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 39 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 39
8.1. Normative References . . . . . . . . . . . . . . . . . . . 39 8.1. Normative References . . . . . . . . . . . . . . . . . . . 39
8.2. Informative References . . . . . . . . . . . . . . . . . . 40 8.2. Informative References . . . . . . . . . . . . . . . . . . 39
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 40 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 40
1. Introduction 1. Introduction
Real time communications over IP networks use SIP for signaling with Real time communications over IP networks use SIP for signaling with
RTP/RTCP for media transport and reporting respectively. These RTP/RTCP for media transport and reporting respectively. These
protocols are very flexible and can support an extremely wide protocols are very flexible and can support an extremely wide
spectrum of usage scenarios. For this reason, extensions to these spectrum of usage scenarios. For this reason, extensions to these
protocols must be specified in the context of a specific usage protocols must be specified in the context of a specific usage
scenario. In this memo, extensions to SIP are proposed to support scenario. In this memo, extensions to SIP are proposed to support
the reporting of RTP Control Protocol Extended Reports [4] metrics. the reporting of RTP Control Protocol Extended Reports [4] metrics.
1.1. Applicability Statement 1.1. Applicability Statement
RTP is utilized in many different architectures and topologies. RFC RTP is utilized in many different architectures and topologies. RFC
5117 [15] lists and describes the following topologies: point to 5117 [13] lists and describes the following topologies: point to
point, point to multipoint using multicast, point to multipoint using point, point to multipoint using multicast, point to multipoint using
the RFC 3550 translator, point to multipoint using the RFC 3550 mixer the RFC 3550 translator, point to multipoint using the RFC 3550 mixer
model, point to multipoint using video switching MCUs, point to model, point to multipoint using video switching MCUs, point to
multipoint using RTCP-terminating MCU, and non-symmetric mixer/ multipoint using RTCP-terminating MCU, and non-symmetric mixer/
translators. As the abstract to this document points out, this translators. As the abstract to this document points out, this
specification is for reporting quality of Voice over Internet specification is for reporting quality of Voice over Internet
Protocol(VoIP) sessions. As such, only the first topology, point to Protocol(VoIP) sessions. As such, only the first topology, point to
point, is currently supported by this specification. This reflects point, is currently supported by this specification. This reflects
both current VoIP deployments which are predominantly point to point both current VoIP deployments which are predominantly point to point
using unicast, and also the state of research in the area of quality. using unicast, and also the state of research in the area of quality.
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The event package supports reporting both the voice quality metrics The event package supports reporting both the voice quality metrics
for both inbound and outbound directions. Voice quality metrics for for both inbound and outbound directions. Voice quality metrics for
the inbound direction can generally be computed locally by the the inbound direction can generally be computed locally by the
reporting endpoint however voice quality metrics for the outbound reporting endpoint however voice quality metrics for the outbound
direction are computed by the remote endpoint and sent to the direction are computed by the remote endpoint and sent to the
reporting endpoint using the RTCP Extended Reports [4]. reporting endpoint using the RTCP Extended Reports [4].
Configuration of usage of the event package is not covered in this Configuration of usage of the event package is not covered in this
document. It is the recommendation of this document that the SIP document. It is the recommendation of this document that the SIP
configuration framework [9] be used. This is discussed in Section configuration framework [15] be used. This is discussed in Section
4.8. 4.8.
The event package SHOULD be used with the SUBSCRIBE/NOTIFY method The event package SHOULD be used with the SUBSCRIBE/NOTIFY method
however it MAY be also used with the PUBLISH method [8] for backward however it MAY be also used with the PUBLISH method [8] for backward
compatibility with some existing implementations. Message flow compatibility with some existing implementations. Message flow
examples for both methods are provided in this document. examples for both methods are provided in this document.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
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quality metric reports using the NOTIFY method. The REPORTER MUST quality metric reports using the NOTIFY method. The REPORTER MUST
NOT send any vq-rtcpxr events if a COLLECTOR address has not been NOT send any vq-rtcpxr events if a COLLECTOR address has not been
configured. The REPORTER populates the Request-URI according to the configured. The REPORTER populates the Request-URI according to the
rules for an in-dialog request. The COLLECTOR MAY send a SUBSCRIBE rules for an in-dialog request. The COLLECTOR MAY send a SUBSCRIBE
to a SIP Proxy acting on behalf of the reporting SIP UA's. to a SIP Proxy acting on behalf of the reporting SIP UA's.
3.2. PUBLISH Method 3.2. PUBLISH Method
A SIP UA that supports this specification MAY also send the service A SIP UA that supports this specification MAY also send the service
quality metric reports using the PUBLISH method [8], however this quality metric reports using the PUBLISH method [8], however this
approach SHOULD NOT be used in unmanaged Internet services. The approach SHOULD NOT be used in general on the public Internet. The
PUBLISH method MAY be supported for backward compatibility with PUBLISH method MAY be supported for backward compatibility with
existing implementations. existing implementations.
The REPORTER MAY therefore populate the Request-URI of the PUBLISH The REPORTER MAY therefore populate the Request-URI of the PUBLISH
method with the address of the COLLECTOR. To ensure security of SIP method with the address of the COLLECTOR. To ensure security of SIP
proxies and the COLLECTOR, the REPORTER MUST be configured with the proxies and the COLLECTOR, the REPORTER MUST be configured with the
address of the COLLECTOR, preferably using the SIP UA configuration address of the COLLECTOR, preferably using the SIP UA configuration
framework [9], as described in section 5.8. framework [15], as described in section 5.8.
It is recommended that the REPORTER send an OPTIONS message to the It is RECOMMENDED that the REPORTER send an OPTIONS message to the
COLLECTOR to ensure support of the PUBLISH message. COLLECTOR to ensure support of the PUBLISH message.
If PUBLISH is not supported, then the reporter can only wait for a
SUBSCRIBE request from the collector and then deliver the
information in NOTIFYs. If a REPORTER sends a PUBLISH to a
COLLECTOR that does not support or allow this method, a 501 Not
Implemented or a 405 Method Not Allowed response will be received,
and the REPORTER will stop publication.
3.3. Multi-Party and Multi-Segment Calls 3.3. Multi-Party and Multi-Segment Calls
A voice quality metric report may be sent for each session A voice quality metric report may be sent for each session
terminating at the REPORTER and may contain multiple report bodies. terminating at the REPORTER and may contain multiple report bodies.
For a multi-party call the report MAY contain report bodies for the For a multi-party call the report MAY contain report bodies for the
session between the reporting endpoint and each remote endpoint for session between the reporting endpoint and each remote endpoint for
which there was an RTP session during the call. which there was an RTP session during the call.
Multi-party services such as call hold and call transfer can result Multi-party services such as call hold and call transfer can result
in the user participating in a series of concatenated sessions, in the user participating in a series of concatenated sessions,
potentially with different choices of codec or sample rate, although potentially with different choices of codec or sample rate, although
these may be perceived by the user as a single call. A REPORTER MAY these may be perceived by the user as a single call. A REPORTER MAY
send a voice quality metric report at the end of each session or MAY send a voice quality metric report at the end of each session or MAY
send a single voice quality metric report containing a report body send a single voice quality metric report containing an application/
for each segment of the call. vq-rtcp-xr body for each segment of the call.
3.4. Overload Avoidance 3.4. Overload Avoidance
Users of this extension should ensure they implement general SIP Users of this extension should ensure they implement general SIP
mechanisms for avoiding overload. For instance, an overloaded proxy mechanisms for avoiding overload. For instance, an overloaded proxy
or COLLECTOR MUST send a 503 Service Unavailable or other 5xx esponse or COLLECTOR MUST send a 503 Service Unavailable or other 5xx
with an appropriate Retry-After time specified. REPORTERs MUST act response with an appropriate Retry-After time specified. REPORTERs
on these responses and respect the retry after time interval. In MUST act on these responses and respect the Retry-After time
addition, future SIP extensions to better handle overload as covered interval. In addition, future SIP extensions to better handle
in [16] should be followed as they are standardized. overload as covered in [14] should be followed as they are
standardized.
To avoid overload of SIP Proxies or COLLECTORS it is important to do To avoid overload of SIP Proxies or COLLECTORS it is important to do
capacity planning and to minimize the number of reports that are capacity planning and to minimize the number of reports that are
sent. sent.
Approaches to avoiding overload include: Approaches to avoiding overload include:
a. Send only one report at the end of each call a. Send only one report at the end of each call
b. Use interval reports only on "problem" calls that are being b. Use interval reports only on "problem" calls that are being
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represent those values as received by the REPORTER. In some represent those values as received by the REPORTER. In some
scenarios, these may not be the same on either end of the session - scenarios, these may not be the same on either end of the session -
the COLLECTOR will need logic to be able to put these sessions the COLLECTOR will need logic to be able to put these sessions
together. The values of parameters such as sample rate, frame together. The values of parameters such as sample rate, frame
duration, frame octets, packets per second, round trip delay, etc duration, frame octets, packets per second, round trip delay, etc
depend on the type of report they are present in. If present in a depend on the type of report they are present in. If present in a
Session or an Interval report, they represent average values over the Session or an Interval report, they represent average values over the
session or interval. If present in an Alert report, they represent session or interval. If present in an Alert report, they represent
instantaneous values. instantaneous values.
The REPORTER always shares local quality reporting information and The REPORTER always includes local quality reporting information and
should, if possible, share remote quality reporting information. should, if possible, share remote quality reporting information to
This remote quality could be available from received RTCP-XR reports the COLLECTOR. This remote quality could be available from received
or other sources. Reporting this is useful in cases where the other RTCP-XR reports or other sources. Reporting this is useful in cases
end might support RTCP-XR but not this voice quality reporting. where the other end might support RTCP-XR but not this voice quality
reporting.
This specification defines a new MIME type application/vq-rtcpxr This specification defines a new MIME type application/vq-rtcpxr
which is a text encoding of the RTCP and RTCP-XR statistics with some which is a text encoding of the RTCP and RTCP-XR statistics with some
additional metrics and correlation information. additional metrics and correlation information.
4.6. Voice Quality Event and Semantics 4.6. Voice Quality Event and Semantics
This section describes the syntax extensions required for event This section describes the syntax extensions required for event
publication in SIP. The formal syntax definitions described in this publication in SIP. The formal syntax definitions described in this
section are expressed in the Augmented BNF [6] format used in SIP section are expressed in the Augmented BNF [6] format used in SIP
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*(WSP Extension) *(WSP Extension)
; RoundTripDelay SHALL be measured as defined in RFC3550 [3]. ; RoundTripDelay SHALL be measured as defined in RFC3550 [3].
RoundTripDelay = "RTD" EQUAL (1*5DIGIT) ;0-65535 RoundTripDelay = "RTD" EQUAL (1*5DIGIT) ;0-65535
; EndSystemDelay metric is defined in RFC 3611 [4] ; EndSystemDelay metric is defined in RFC 3611 [4]
EndSystemDelay = "ESD" EQUAL (1*5DIGIT) ;0-65535 EndSystemDelay = "ESD" EQUAL (1*5DIGIT) ;0-65535
; OneWayDelay is defined in RFC 2679 [14] ; OneWayDelay is defined in RFC 2679 [12]
OneWayDelay = "OWD" EQUAL (1*5DIGIT) ;0-65535 OneWayDelay = "OWD" EQUAL (1*5DIGIT) ;0-65535
; SymmOneWayDelay is defined as half the sum of RoundTripDelay ; SymmOneWayDelay is defined as half the sum of RoundTripDelay
; and the EndSystemDelay values for both endpoints. ; and the EndSystemDelay values for both endpoints.
SymmOneWayDelay = "SOWD" EQUAL (1*5DIGIT); 0-65535 SymmOneWayDelay = "SOWD" EQUAL (1*5DIGIT); 0-65535
; Interarrival Jitter is calculated as defined RFC 3550 ; Interarrival Jitter is calculated as defined RFC 3550
; and converted into milliseconds ; and converted into milliseconds
InterarrivalJitter = "IAJ" EQUAL (1*5DIGIT) ;0-65535 ms InterarrivalJitter = "IAJ" EQUAL (1*5DIGIT) ;0-65535 ms
; Mean Absolute Jitter is measured as defined ; Mean Absolute Jitter is measured as defined
; by ITU-T G.1020 [11] where it is known as MAPDV ; by ITU-T G.1020 [9] where it is known as MAPDV
MeanAbsoluteJitter = "MAJ" EQUAL (1*5DIGIT);0-65535 MeanAbsoluteJitter = "MAJ" EQUAL (1*5DIGIT);0-65535
; Signal metrics definitions are provided in RFC 3611 ; Signal metrics definitions are provided in RFC 3611
Signal = "Signal" HCOLON Signal = "Signal" HCOLON
[ SignalLevel WSP ] [ SignalLevel WSP ]
[ NoiseLevel WSP ] [ NoiseLevel WSP ]
[ ResidualEchoReturnLoss ] [ ResidualEchoReturnLoss ]
*(WSP Extension) *(WSP Extension)
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[ ExtROutEstAlg WSP ] [ ExtROutEstAlg WSP ]
[ MOS-LQ WSP ] [ MOS-LQ WSP ]
[ MOSLQEstAlg WSP ] [ MOSLQEstAlg WSP ]
[ MOS-CQ WSP ] [ MOS-CQ WSP ]
[ MOSCQEstAlg WSP ] [ MOSCQEstAlg WSP ]
[ QoEEstAlg ] [ QoEEstAlg ]
*(WSP Extension) *(WSP Extension)
ListeningQualityR = "RLQ" EQUAL (1*3DIGIT) ; 0 - 120 ListeningQualityR = "RLQ" EQUAL (1*3DIGIT) ; 0 - 120
RLQEstAlg = "RLQEstAlg" EQUAL word ; "P.564" [12], or other RLQEstAlg = "RLQEstAlg" EQUAL word ; "P.564" [10], or other
ConversationalQualityR = "RCQ" EQUAL (1*3DIGIT) ; 0 - 120 ConversationalQualityR = "RCQ" EQUAL (1*3DIGIT) ; 0 - 120
RCQEstAlg = "RCQEstAlg" EQUAL word ; "P.564", or other RCQEstAlg = "RCQEstAlg" EQUAL word ; "P.564", or other
; ExternalR-In is measured by the local endpoint for incoming ; ExternalR-In is measured by the local endpoint for incoming
; connection on "other" side of this endpoint ; connection on "other" side of this endpoint
; e.g. Phone A <---> Bridge <----> Phone B ; e.g. Phone A <---> Bridge <----> Phone B
; ListeningQualityR = quality for Phone A ----> Bridge path ; ListeningQualityR = quality for Phone A ----> Bridge path
; ExternalR-In = quality for Bridge <---- Phone B path ; ExternalR-In = quality for Bridge <---- Phone B path
ExternalR-In = "EXTRI" EQUAL (1*3DIGIT) ; 0 - 120 ExternalR-In = "EXTRI" EQUAL (1*3DIGIT) ; 0 - 120
ExtRInEstAlg = "ExtRIEstAlg" EQUAL word ; "P.564" or other ExtRInEstAlg = "ExtRIEstAlg" EQUAL word ; "P.564" or other
; ExternalR-Out is copied from RTCP XR message received from the ; ExternalR-Out is copied from RTCP XR message received from the
; remote endpoint on "other" side of this endpoint ; remote endpoint on "other" side of this endpoint
; e.g. Phone A <---> Bridge <----> Phone B ; e.g. Phone A <---> Bridge <----> Phone B
; ExternalR-Out = quality for Bridge -----> Phone B path ; ExternalR-Out = quality for Bridge -----> Phone B path
ExternalR-Out = "EXTRO" EQUAL (1*3DIGIT) ; 0 - 120 ExternalR-Out = "EXTRO" EQUAL (1*3DIGIT) ; 0 - 120
ExtROutEstAlg = "ExtROEstAlg" EQUAL word ; "P.564" or other ExtROutEstAlg = "ExtROEstAlg" EQUAL word ; "P.564" or other
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This parameter is not contained in RTP or SDP but can usually be This parameter is not contained in RTP or SDP but can usually be
obtained from the device codec. This field provides the number of obtained from the device codec. This field provides the number of
frames in each RTP packet. Note this value can be combined with the frames in each RTP packet. Note this value can be combined with the
FrameDuration to determine the packetization rate. Also, where a FrameDuration to determine the packetization rate. Also, where a
sample-based codec is used, a "frame" refers to the set of samples sample-based codec is used, a "frame" refers to the set of samples
carried in an RTP packet. carried in an RTP packet.
4.6.2.3.8. FMTP Options 4.6.2.3.8. FMTP Options
This parameter is taken directly from the SDP attribute "fmtp". This parameter is taken directly from the SDP attribute "fmtp"
defined in RFC4566.
4.6.2.3.9. Silence Suppression State 4.6.2.3.9. Silence Suppression State
This parameter does not correspond to SDP, RTP, or RTCP XR. It This parameter does not correspond to SDP, RTP, or RTCP XR. It
indicates whether silence suppression, also known as Voice Activity indicates whether silence suppression, also known as Voice Activity
Detection (VAD) is enabled for the identified session. Detection (VAD) is enabled for the identified session.
4.6.2.3.10. Packet Loss Concealment 4.6.2.3.10. Packet Loss Concealment
This value corresponds to "PLC" in RFC3611 in the VoIP Metrics Report This value corresponds to "PLC" in RFC3611 in the VoIP Metrics Report
Block. The values defined by RFC3611 are reused by this Block. The values defined by RFC3611 are reused by this
recommendation and therefore no mapping is required. recommendation and therefore no mapping is required.
4.6.2.4. LocalAddr 4.6.2.4. LocalAddr
This field provides the IP address, port and synchronization source This field provides the IP address, port and synchronization source
(SSRC) for the session from the perspective of the endpoint that is (SSRC) for the session from the perspective of the endpoint that is
measuring performance. The IPAddress can be IPv4 or IPv6 format. measuring performance. The IPAddress MAY be IPv4 or IPv6 format.
The SSRC is taken from SDP, RTCP, or RTCP XR input parameters. The SSRC is taken from SDP, RTCP, or RTCP XR input parameters.
In the presence of NAT and where a NAT-traversal mechanism such as In the presence of NAT and where a NAT-traversal mechanism such as
STUN [10] is used, the external IP address can be reported, since the STUN [16] is used, the external IP address can be reported, since the
internal IP address is not visible to the network operator. internal IP address is not visible to the network operator.
4.6.2.5. RemoteAddr 4.6.2.5. RemoteAddr
This field provides the IP address, port and ssrc of the session peer This field provides the IP address, port and ssrc of the session peer
from the perspective of the remote endpoint measuring performance. from the perspective of the remote endpoint measuring performance.
In the presence of NAT and where a NAT-traversal mechanism such as In the presence of NAT and where a NAT-traversal mechanism such as
STUN [10] is used, the external IP address can be reported, since the STUN [16] is used, the external IP address can be reported, since the
internal IP address is not visible to the network operator. internal IP address is not visible to the network operator.
4.6.2.6. Jitter Buffer Parameters 4.6.2.6. Jitter Buffer Parameters
4.6.2.6.1. Jitter Buffer Adaptive 4.6.2.6.1. Jitter Buffer Adaptive
This value corresponds to "JBA" in RFC3611 in the VoIP Metrics Report This value corresponds to "JBA" in RFC3611 in the VoIP Metrics Report
Block. The values defined by RFC3611 are unchanged and therefore no Block. The values defined by RFC3611 are unchanged and therefore no
mapping is required. mapping is required.
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Metrics Report Block. This value requires no conversion; the exact Metrics Report Block. This value requires no conversion; the exact
value sent in an RTCP XR VoIP Metrics Report Block can be reported. value sent in an RTCP XR VoIP Metrics Report Block can be reported.
4.6.2.8.5. Minimum Gap Threshold 4.6.2.8.5. Minimum Gap Threshold
This value corresponds to "Gmin" in RFC3611 in the VoIP Metrics This value corresponds to "Gmin" in RFC3611 in the VoIP Metrics
Report Block. This value requires no conversion; the exact value Report Block. This value requires no conversion; the exact value
sent in an RTCP XR VoIP Metrics Report Block can be reported. sent in an RTCP XR VoIP Metrics Report Block can be reported.
4.6.2.9. Delay Parameters 4.6.2.9. Delay Parameters
4.6.2.9.1. Round Trip Delay 4.6.2.9.1. Round Trip Delay
This value corresponds to "round trip delay" in RFC3611 in the VoIP This value corresponds to "round trip delay" in RFC3611 in the VoIP
Metrics Report Block and may be measured using the method defined in Metrics Report Block and may be measured using the method defined in
RFC3550. The parameter is expressed in milliseconds. RFC3550. The parameter is expressed in milliseconds.
4.6.2.9.2. End System Delay 4.6.2.9.2. End System Delay
This value corresponds to "end system delay" in RFC3611 in the VoIP This value corresponds to "end system delay" in RFC3611 in the VoIP
Metrics Report Block. This parameter does not require any Metrics Report Block. This parameter does not require any
conversion. The parameter is expressed in milliseconds. conversion. The parameter is expressed in milliseconds.
4.6.2.9.3. Symmetric One Way Delay 4.6.2.9.3. Symmetric One Way Delay
This value is computed by adding Round Trip Delay to the local and This value is computed by adding Round Trip Delay to the local and
remote End System Delay and dividing by two. remote End System Delay and dividing by two.
4.6.2.9.4. One Way Delay 4.6.2.9.4. One Way Delay
This value SHOULD be measured using the methods defined in IETF RFC This value SHOULD be measured using the methods defined in IETF RFC
2679 [14]. The parameter is expressed in milliseconds. 2679 [12]. The parameter is expressed in milliseconds.
4.6.2.9.5. Inter-arrival Jitter 4.6.2.9.5. Inter-arrival Jitter
Inter-arrival jitter is calculated as defined in RFC 3550 and Inter-arrival jitter is calculated as defined in RFC 3550 and
converted into milliseconds. converted into milliseconds.
4.6.2.9.6. Mean Absolute Jitter 4.6.2.9.6. Mean Absolute Jitter
It is recommended that MAJ be measured as defined in ITU-T G.1020 It is recommended that MAJ be measured as defined in ITU-T G.1020
[11]. This parameter is often referred to as MAPDV. The parameter [9]. This parameter is often referred to as MAPDV. The parameter is
is expressed in milliseconds. expressed in milliseconds.
4.6.2.10. Signal-related Parameters 4.6.2.10. Signal-related Parameters
4.6.2.10.1. Signal Level 4.6.2.10.1. Signal Level
This field corresponds to "signal level" in RFC3611 in the VoIP This field corresponds to "signal level" in RFC3611 in the VoIP
Metrics Report Block. This field provides the voice signal relative Metrics Report Block. This field provides the voice signal relative
level is defined as the ratio of the signal level to a 0 dBm0 level is defined as the ratio of the signal level to a 0 dBm0
reference, expressed in decibels. This value can be used directly reference, expressed in decibels. This value can be used directly
without extra conversion. without extra conversion.
skipping to change at page 26, line 22 skipping to change at page 26, line 38
directly without extra conversion. directly without extra conversion.
4.6.2.11. Quality Scores 4.6.2.11. Quality Scores
4.6.2.11.1. ListeningQualityR 4.6.2.11.1. ListeningQualityR
This field reports the listening quality expressed as an R factor This field reports the listening quality expressed as an R factor
(per G.107). This does not include the effects of echo or delay. (per G.107). This does not include the effects of echo or delay.
The range of R is 0-95 for narrowband calls and 0-120 for wideband The range of R is 0-95 for narrowband calls and 0-120 for wideband
calls. Algorithms for computing this value SHOULD be compliant with calls. Algorithms for computing this value SHOULD be compliant with
ITU-T Recommendations P.564 [12] and G.107 [13]. ITU-T Recommendations P.564 [10] and G.107 [11].
4.6.2.11.2. RLQEstAlg 4.6.2.11.2. RLQEstAlg
This field provides a text name for the algorithm used to estimate This field provides a text name for the algorithm used to estimate
ListeningQualityR. This field will be free form text and not ListeningQualityR. This field will be free form text and not
necessarily reflective of any standards or recommendations. necessarily reflective of any standards or recommendations.
4.6.2.11.3. ConversationalQualityR 4.6.2.11.3. ConversationalQualityR
This field corresponds to "R factor" in RFC3611 in the VoIP Metrics This field corresponds to "R factor" in RFC3611 in the VoIP Metrics
skipping to change at page 28, line 5 skipping to change at page 28, line 20
MOS-xQ parameter if the RFC3611 reported value is 127 (which MOS-xQ parameter if the RFC3611 reported value is 127 (which
indicates unavailable). indicates unavailable).
4.6.2.11.9.1. MOS-LQ 4.6.2.11.9.1. MOS-LQ
This field corresponds to "MOSLQ" in RFC3611 in the VoIP Metrics This field corresponds to "MOSLQ" in RFC3611 in the VoIP Metrics
Report Block. This parameter is the estimated mean opinion score for Report Block. This parameter is the estimated mean opinion score for
listening voice quality on a scale from 1 to 5, in which 5 represents listening voice quality on a scale from 1 to 5, in which 5 represents
"Excellent" and 1 represents "Unacceptable". Algorithms for "Excellent" and 1 represents "Unacceptable". Algorithms for
computing this value SHOULD be compliant with ITU-T Recommendation computing this value SHOULD be compliant with ITU-T Recommendation
P.564 [12]. This field provides a text name for the algorithm used P.564 [10]. This field provides a text name for the algorithm used
to estimate MOS-LQ. to estimate MOS-LQ.
4.6.2.11.9.2. MOS-CQ 4.6.2.11.9.2. MOS-CQ
This field corresponds to "MOSCQ" in RFC3611 in the VoIP Metrics This field corresponds to "MOSCQ" in RFC3611 in the VoIP Metrics
Report Block. This parameter is the estimated mean opinion score for Report Block. This parameter is the estimated mean opinion score for
conversation voice quality on a scale from 1 to 5, in which 5 conversation voice quality on a scale from 1 to 5, in which 5
represents excellent and 1 represents unacceptable. Algorithms for represents excellent and 1 represents unacceptable. Algorithms for
computing this value SHOULD be compliant with ITU-T Recommendation computing this value SHOULD be compliant with ITU-T Recommendation
P.564 with regard to the listening quality element of the computed P.564 with regard to the listening quality element of the computed
skipping to change at page 30, line 15 skipping to change at page 30, line 29
SUBSCRIBE, NOTIFY SUBSCRIBE, NOTIFY
Event: vq-rtcpxr Event: vq-rtcpxr
Accept: application/sdp, message/sipfrag Accept: application/sdp, message/sipfrag
Subscription-State: active;expires=3600 Subscription-State: active;expires=3600
Content-Type: application/vq-rtcpxr Content-Type: application/vq-rtcpxr
Content-Length: ... Content-Length: ...
VQSessionReport: CallTerm VQSessionReport: CallTerm
CallID: 6dg37f1890463 CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org> LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@elpmaxe.org> RemoteID: Bill <sip:bill@example.net>
OrigID: Alice <sip:alice@example.org> OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671 LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871 RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02 RemoteMAC: 00:26:08:8e:95:02
LocalMetrics: LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50 SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
skipping to change at page 34, line 31 skipping to change at page 34, line 45
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY SUBSCRIBE, NOTIFY
Event: vq-rtcpxr Event: vq-rtcpxr
Accept: application/sdp, message/sipfrag Accept: application/sdp, message/sipfrag
Content-Type: application/vq-rtcpxr Content-Type: application/vq-rtcpxr
Content-Length: ... Content-Length: ...
VQSessionReport: CallTerm VQSessionReport: CallTerm
CallID: 6dg37f1890463 CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org> LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@elpmaxe.org> RemoteID: Bill <sip:bill@example.net>
OrigID: Alice <sip:alice@example.org> OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671 LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871 RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02 RemoteMAC: 00:26:08:8e:95:02
LocalMetrics: LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50 SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
skipping to change at page 37, line 15 skipping to change at page 37, line 29
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10 Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-23 NL=-60 RERL=55 Signal:SL=-23 NL=-60 RERL=55
QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3 QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3
QoEEstAlg=P.564 QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761; DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh3111 from-tag=9123dh3111
4.8. Configuration Dataset for vq-rtcpxr Events 4.8. Configuration Dataset for vq-rtcpxr Events
It is the suggestion of the authors that the SIP configuration It is the suggestion of the authors that the SIP configuration
framework [9] be used to establish the necessary parameters for usage framework [15] be used to establish the necessary parameters for
of vq-rtcpxr events. A dataset for this purpose should be designed usage of vq-rtcpxr events. A dataset for this purpose should be
and documented in a separate draft upon completion of the framework. designed and documented in a separate draft upon completion of the
framework.
5. IANA Considerations 5. IANA Considerations
This document registers a new SIP Event Package and a new MIME type. This document registers a new SIP Event Package and a new MIME type.
5.1. SIP Event Package Registration 5.1. SIP Event Package Registration
Package name: vq-rtcpx Package name: vq-rtcpx
Type: package Type: package
Contact: Amy Pendleton <aspen@telchemy.com> Contact: Amy Pendleton <aspen@telchemy.com>
skipping to change at page 39, line 32 skipping to change at page 39, line 32
[6] Crocker, D. and P. Overell, "Augmented BNF for Syntax [6] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008. Specifications: ABNF", STD 68, RFC 5234, January 2008.
[7] Klyne, G., Ed. and C. Newman, "Date and Time on the Internet: [7] Klyne, G., Ed. and C. Newman, "Date and Time on the Internet:
Timestamps", RFC 3339, July 2002. Timestamps", RFC 3339, July 2002.
[8] Niemi, A., "Session Initiation Protocol (SIP) Extension for [8] Niemi, A., "Session Initiation Protocol (SIP) Extension for
Event State Publication", RFC 3903, October 2004. Event State Publication", RFC 3903, October 2004.
[9] Channabasappa, S., "A Framework for Session Initiation Protocol [9] ITU-T G.1020, "Performance parameter definitions for quality of
User Agent Profile Delivery",
draft-ietf-sipping-config-framework-17 (work in progress),
February 2010.
[10] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session
Traversal Utilities for NAT (STUN)", RFC 5389, October 2008.
[11] ITU-T G.1020, "Performance parameter definitions for quality of
speech and other voiceband applications utilizing IP speech and other voiceband applications utilizing IP
networks.". networks.".
[12] ITU-T P.564, "Conformance testing for voice over IP [10] ITU-T P.564, "Conformance testing for voice over IP
transmission quality assessment models.". transmission quality assessment models.".
[13] ITU-T G.107, "The E-model, a computational model for use in [11] ITU-T G.107, "The E-model, a computational model for use in
transmission planning.". transmission planning.".
[14] Almes, G., Kalidindi, S., and M. Zekauskas, "A One-way Delay [12] Almes, G., Kalidindi, S., and M. Zekauskas, "A One-way Delay
Metric for IPPM", RFC 2679, September 1999. Metric for IPPM", RFC 2679, September 1999.
8.2. Informative References 8.2. Informative References
[15] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, [13] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
January 2008. January 2008.
[16] Hilt, V., Noel, E., Shen, C., and A. Abdelal, "Design [14] Hilt, V., Noel, E., Shen, C., and A. Abdelal, "Design
Considerations for Session Initiation Protocol (SIP) Overload Considerations for Session Initiation Protocol (SIP) Overload
Control", draft-ietf-sipping-overload-design-02 (work in Control", draft-ietf-sipping-overload-design-02 (work in
progress), July 2009. progress), July 2009.
[15] Channabasappa, S., "A Framework for Session Initiation Protocol
User Agent Profile Delivery",
draft-ietf-sipping-config-framework-17 (work in progress),
February 2010.
[16] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session
Traversal Utilities for NAT (STUN)", RFC 5389, October 2008.
Authors' Addresses Authors' Addresses
Amy Pendleton Amy Pendleton
Telchemy Incorporated Telchemy Incorporated
Email: aspen@telchemy.com Email: aspen@telchemy.com
Alan Clark Alan Clark
Telchemy Incorporated Telchemy Incorporated
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