Internet Draft K. Hedayat
Expires: January 28, 2009 Brix Networks
N. Venna
Brix Networks
P. Jones
Cisco Systems, Inc.
A. Roychowdhury
Hughes Systique Corp.
C. SivaChelvan
Cisco Systems, Inc.
N. Stratton
BlinkMind, Inc.
July 28, 2008
An Extension to the Session Description Protocol (SDP) for Media
Loopback
draft-ietf-mmusic-media-loopback-09
Status of this Memo
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Copyright Notice
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Copyright (C) The IETF Trust (2008).
Abstract
The wide deployment of Voice over IP (VoIP), Real-time Text and
Video over IP services has introduced new challenges in managing
and maintaining voice/real-time Text/video quality, reliability,
and overall performance. In particular, media delivery is an area
that needs attention. One method of meeting these challenges is
monitoring the media delivery performance by looping media back to
the transmitter. This is typically referred to as "active
monitoring" of services. Media loopback is especially popular in
ensuring the quality of transport to the edge of a given VoIP,
Real-time Text or Video over IP service. Today in networks that
deliver real-time media, short of running 'ping' and 'traceroute'
to the edge, service providers are left without the necessary tools
to actively monitor, manage, and diagnose quality issues with their
service. The extension defined herein adds new SDP media
attributes which enables establishment of media sessions where the
media is looped back to the transmitter. Such media sessions will
serve as monitoring and troubleshooting tools by providing the
means for measurement of more advanced VoIP, Real-time Text and
Video Over IP performance metrics.
Table of Contents
1. Introduction..................................................3
2. Terminology...................................................4
3. Offering Entity Behavior......................................4
4. Answering Entity Behavior.....................................4
5. SDP Constructs Syntax.........................................4
5.1 Loopback Type Attribute...................................4
5.2 Loopback Mode Attribute...................................6
5.3 Generating the Offer for Loopback Session.................7
5.4 Generating the Answer for Loopback Session................8
5.5 Offerer Processing of the Answer..........................9
5.6 Modifying the Session....................................10
6. RTP Requirements.............................................10
7. Payload formats for Packet loopback..........................10
7.1 Encapsulated Payload format..............................11
7.2 Direct loopback RTP payload format.......................13
8. RTCP Requirements............................................14
9. Congestion Control...........................................15
10. Examples....................................................15
10.1 Offer for specific media loopback type..................15
10.2 Offer for choice of media loopback type.................16
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10.3 Offer for choice of media loopback type with
rtp-start-loopback...........................................17
10.4 Response to INVITE request rejecting loopback media.....18
10.5 Response to INVITE request rejecting loopback media with
rtp-start-loopback...........................................18
11. Security Considerations.....................................19
12. Implementation Considerations...............................20
13. IANA Considerations.........................................20
13.1 SDP Attributes..........................................20
13.2 MIME Types..............................................21
14. Acknowledgements............................................30
15. Normative References........................................30
1. Introduction
The overall quality, reliability, and performance of VoIP,
Real-time Text and Video over IP services rely on the performance
and quality of the media path. In order to assure the quality of
the delivered media there is a need to monitor the performance of
the media transport. One method of monitoring and managing the
overall quality of VoIP, Real-time Text and Video over IP Services
is through monitoring the quality of the media in an active
session. This type of "active monitoring" of services is a method
of pro-actively managing the performance and quality of VoIP based
services.
The goal of active monitoring is to measure the media quality of a
VoIP, Real-time Text or Video over IP session. A way to achieve
this goal is to request an endpoint to loop media back to the other
endpoint and to provide media statistics (e.g., RTCP and RTCP XR
information). Another method involves deployment of special
endpoints that always loop incoming media back for sessions.
Although the latter method has been used and is functional, it does
not scale to support large networks and introduces new network
management challenges. Further, it does not offer the granularity
of testing a specific endpoint that may be exhibiting problems.
The extension defined in this memo introduces new SDP media
attributes that enable establishment of media sessions where the
media is looped back to the transmitter. The offer/answer model
[RFC3264] is used to establish a loopback connection. Furthermore,
this extension provides guidelines on handling RTP [RFC3550], as
well as usage of RTCP [RFC3550] and RTCP XR [RFC3611] for reporting
media related measurements.
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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 RFC 2119.
3. Offering Entity Behavior
An offering entity compliant to this memo and attempting to
establish a media session with media loopback MUST include
"loopback" media attributes for each individual media description
in the offer message. The offering entity MUST look for the
"loopback" media attributes in the media description(s) of the
response from the answering entity for confirmation that the
request is accepted.
4. Answering Entity Behavior
An answering entity compliant to this specification and receiving
an offer containing media descriptions with the "loopback" media
attributes, MUST acknowledge the request by including the received
"loopback" media attributes for each media description in its
response. The server MAY reject the "loopback" request for
specific media types as defined in section 5.4.1 of this
specification.
An answering entity that is not compliant to this specification and
which receives an offer with the "loopback" media attributes MAY
ignore the attribute and treat the incoming offer as a normal
request.
5. SDP Constructs Syntax
Two new media attributes are defined: one indicates the type of
loopback and one indicates the mode of the loopback.
5.1 Loopback Type Attribute
The loopback type is a property media attribute with the following
syntax:
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a=loopback:<loopback-type>
Following is the Augmented BNF [RFC4234] for loopback-type:
loopback-type = loopback-type-choices | loopback-type-choice-3
loopback-choices = loopback-type-choice-1 | loopback-type-choice-2
| loopback-type-choice-1 1*space loopback-type-choice-2 |
loopback-type-choice-2 1*space loopback-type-choice-1
loopback-type-choice-1 = "rtp-pkt-loopback"
loopback-type-choice-2 = "rtp-media-loopback"
loopback-type-choice-3 = "rtp-start-loopback"
The loopback type is used to indicate the type of loopback. The
loopback-type values are rtp-pkt-loopback, rtp-media-loopback, and
rtp-start-loopback.
rtp-pkt-loopback: In this mode, the RTP packets are looped back to
the sender at a point before the encoder/decoder function in the
receive direction to a point after the encoder/decoder function in
the send direction. This effectively re-encapsulates the RTP
payload with the RTP/UDP/IP overheads appropriate for sending it in
the reverse direction. Any type of encoding related functions,
such as packet loss concealment, MUST NOT be part of this type of
loopback path. In this mode the RTP packets are looped back with a
new payload type and format. Section 7 describes the payload
formats that MUST be used for this type of loopback.
rtp-media-loopback: This loopback is activated as close as possible
to the analog interface and after the decoder so that the RTP
packets are subsequently re-encoded prior to transmission back to
the sender.
rtp-start-loopback: In certain scenarios it is possible that the
media transmitted by the loopback-source is blocked by a network
element until the loopback-mirror starts transmitting packets.
Loopback-source and loopback-mirror are loopback modes defined in
section 5.2. One example of this scenario is the presence of an
RTP relay in the path of the media. RTP relays exist in VoIP
networks for purpose of NAT and Firewall traversal. If an RTP
relay is present, the loopback-source's packets are dropped by the
RTP relay until the loopback-mirror has started transmitting media
and the media state within the RTP relay is established. This
loopback attribute is used to specify the media type for
transmitting media packets by the loopback-mirror prior to the
loopback process for the purpose of setting media state within the
network. In the presence of this loopback attribute the loopback-
mirror will transmit media, according to the description that
contains this attribute, until it receives media from the loopback-
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source. The loopback-mirror MAY include this attribute in the
answer if it is not present in the offer. This may be necessary if
the loopback-mirror is aware of NAT's, firewalls, or RTP relays on
the path of the call. In this case the loopback-source MUST accept
media according to rtp-start-loopback attribute. After the first
media packet is received from the loopback-source, the loopback-
mirror MUST terminate the transmission of rtp-start-loopback media
and MUST start looping back media as defined by the other loopback
attributes present in the offer. If an offer includes the
rtp-start-loopback attribute it MUST also include at least one
other attribute as defined in this section. The loopback-source is
able to filter rtp-start-loopback packets from other types of
loopback with the payload type of the packet. The media port number
for rtp-start-loopback MUST be the same as the corresponding
loopback attribute that will take over after the reception of first
media packet from the offering entity.
It is recommended that an offering entity specifying media with
either rtp-pkt-loopback or rtp-media-loopback attribute also
specify the rtp-start-loopback attribute unless the offering entity
is certain that its media will not be blocked by a network entity
as explained above.
5.2 Loopback Mode Attribute
The loopback mode is a value media attribute that is used to
indicate the mode of the loopback. These attributes are additional
mode attributes like sendonly, recvonly, etc. The syntax of the
loopback mode media attribute is:
a=<loopback-mode>:<fmt>...
The loopback-mode values are loopback-source and loopback-mirror.
loopback-source: This attribute specifies that the sender is the
media source and expects the receiver to act as a loopback-mirror.
loopback-mirror: This attribute specifies that the receiver will
mirror (echo) all received media back to the sender of the RTP
stream. No media is generated locally by the receiver for
transmission in the mirrored stream unless rtp-start-loopback is
requested.
<fmt> is a media format description. The format descrption has the
semantics as defined in section 5.14 of RFC 4566[RFC4566]. When
loopback-mode is specified as loopback-source, the media format
corresponds to the RTP payload types the source is willing to send.
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When loopback-mode is specified as loopback-mirror, the media
format corresponds to the RTP payload types the mirror is willing
to receive. The payload types specified in m= line for a
loopback-source specify the payloads the source is willing to
receive. Similarly, for the loopback-mirror these payload types
specify the payloads it is willing to send.
The loopback mode attribute does not apply to rtp-start-loopback
attribute and MUST be ignored if received by the answering entity.
5.3 Generating the Offer for Loopback Session
If an offerer wishes to make a loopback request, it MUST include
both the loopback-type and loopback-mode attribute in a valid SDP
offer:
Example: m=audio 41352 RTP/AVP 0 8
a=loopback:rtp-media-loopback
a=loopback-source:0 8
Note: A loopback offer in a given media description MUST NOT
contain the standard mode attributes sendonly, recvonly, sendrecv
or inactive. The loopback-mode attributes (loopback-source and
loopback-mirror) replace the standard attributes.
The offerer may offer more than one loopback-type in the SDP offer.
In this case the answer MUST include only one of the loopback types
that are accepted by the answerer. The answerer SHOULD give
preference to the first loopback-type in the SDP offer.
For loopback-source media (e.g. audio) streams, the port number and
the address in the offer (m= line) indicate where the offerer would
like to receive the media stream. The payload type numbers
indicate the value of the payload the offerer expects to receive.
The RTP payload types indicated in the a=loopback-source line are
the payload types for the codecs the offerer is willing to send.
However, the answer might indicate a different payload type number
for the same codec. In that case, the offerer MUST send the
payload type received in the answer.
If loopback-type is rtp-pkt-loopback, the loopback-mirror MUST send
and the loopback-source MUST receive the looped back packets
encoded in one of the two payload formats (encapsulated RTP or
payload loopback) as defined in section 7.
Example: m=audio 41352 RTP/AVP 112
a=loopback:rtp-pkt-loopback
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a=loopback-source:0 8
a=rtpmap:112 encaprtp/8000
Example: m=audio 41352 RTP/AVP 112
a=loopback:rtp-pkt-loopback
a=loopback-source:0 8
a=rtpmap:112 rtploopback/8000
Note: NAT devices may change the actual port number that is used
for transmission and the expected receive port.
5.4 Generating the Answer for Loopback Session
If an answerer wishes to accept the loopback request it MUST
include both the loopback mode and loopback type attribute in the
answer. If a stream is offered with loopback-source or
loopback-mirror attributes, the corresponding stream MUST be
loopback-mirror or loopback-source respectively, provided that
answerer is capable of supporting the requested loopback-type.
For example, if the offer contains:
m=audio 41352 RTP/AVP 0 8
a=loopback:rtp-media-loopback
a=loopback-source:0 8
The answer that is capable of supporting the offer MUST contain:
m=audio 41352 RTP/AVP 0 8
a=loopback:rtp-media-loopback
a=loopback-mirror:0 8
As previously stated if a stream is offered with multiple loopback
type attributes, the corresponding stream MUST contain only one
loopback type attribute selected by the answerer.
For example, if the offer contains:
m=audio 41352 RTP/AVP 0 8 112
a=loopback:rtp-media-loopback rtp-pkt-loopback
a=loopback-source:0 8
The answer that is capable of supporting the offer and chooses to
loopback the media using the rtp-media-loopback type MUST contain:
m=audio 41352 RTP/AVP 0 8
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a=loopback:rtp-media-loopback
a=loopback-mirror:0 8
As specified in section 7, if the loopback-type is
rtp-pkt-loopback, either the encapsulated RTP payload format or
direct loopback RTP payload format MUST be used for looped back
packets.
For example, if the offer contains:
m=audio 41352 RTP/AVP 112 113
a=loopback:rtp-pkt-loopback
a=loopback-source:0 8
a=rtpmap:112 encaprtp/8000
a=rtpmap:113 rtploopback/8000
The answer that is capable of supporting the offer MUST contain one
of the following:
m=audio 41352 RTP/AVP 112
a=loopback:rtp-pkt-loopback
a=loopback-mirror:0 8
a=rtpmap:112 encaprtp/8000
m=audio 41352 RTP/AVP 113
a=loopback:rtp-pkt-loopback
a=loopback-mirror:0 8
a=rtpmap:113 rtploopback/8000
5.4.1 Rejecting the Loopback Offer
An offered stream with loopback-source MAY be rejected if the
loopback-type is not specified, the specified loopback-type is not
supported, or the endpoint cannot honor the offer for any other
reason. The Loopback request may be rejected by setting the media
port number to zero in the answer as per RFC 3264 [RFC3264].
5.5 Offerer Processing of the Answer
The answer to a loopback-source MUST be loopback-mirror. The
answer to a loopback-mirror MUST be loopback-source. The loopback-
mode line MUST contain at least one codec the answerer is willing
to send or receive depending on whether it is the loopback-source
or the loopback-mirror. In addition, the "m=" line MUST contain at
least one codec that the answerer is willing to send or receive
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depending on whether it is the loopback-mirror or the loopback-
source.
If the answer does not contain a=loopback-mirror or
a=loopback-source or contains any other standard mode attributes,
it is assumed that the loopback extensions are not supported by the
target UA.
5.6 Modifying the Session
At any point during the loopback session, either participant may
issue a new offer to modify the characteristics of the previous
session. In case of SIP this is defined in section 8 of RFC 3264
[RFC3264]. This also includes transitioning from a normal media
processing mode to loopback mode, and vice a versa.
6. RTP Requirements
An answering entity that is compliant to this specification and
accepting a media with rtp-pkt-loopback loopback-type MUST loopback
the incoming RTP packets using either the encapsulated RTP payload
format or the direct loopback RTP payload format as defined in
section 7 of this specification.
An answering entity that is compliant to this specification and
accepting a media with rtp-media-loopback loopback-type MUST
transmit all received media back to the sender. The incoming media
MUST be treated as if it were to be played (e.g. the media stream
MAY receive treatment from PLC algorithms). The answering entity
MUST re-generate all the RTP header fields as it would when
transmitting media. The answering entity MAY choose to encode the
loopback media according to any of the media descriptions supported
by the offering entity. Furthermore, in cases where the same media
type is looped back, the answering entity MAY choose to preserve
number of frames/packet and bitrate of the encoded media according
to the received media.
7. Payload formats for Packet loopback
The payload formats described in this section MUST be used by a
loopback-mirror when rtp-pkt-loopback is the specified
loopback-type. Two different formats are specified here - an
encapsulated RTP payload format and a direct loopback RTP payload
format. The encapsulated RTP payload format should be used when
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the incoming RTP header information needs to be preserved during
the loopback operation. This is useful in cases where loopback
source needs to measure performance metrics in both directions.
However, this comes at the expense of increased packet size as
described in section 7.1. The direct loopback RTP payload format
should be used when bandwidth requirement prevent the use of
encapsulated RTP payload format.
7.1 Encapsulated Payload format
A received RTP packet is encapsulated in the payload section of the
RTP packet generated by a loopback-mirror. Each received packet
MUST be encapsulated in a different packet, the encapsulated packet
MAY be fragmented only if required (for example: due to MTU
limitations).
7.1.1 Usage of RTP Header fields
Payload Type (PT): The assignment of an RTP payload type for this
packet format is outside the scope of this document; it is either
specified by the RTP profile under which this payload format is
used or more likely signaled dynamically out-of-band (e.g., using
SDP; section 7.1.3 defines the name binding).
Marker (M) bit: If the received RTP packet is looped back in
multiple RTP packets, the M bit is set to 1 in the last packet,
otherwise it is set to 0.
Extension (X) bit: Defined by the RTP Profile used.
Sequence Number: The RTP sequence number SHOULD be generated by the
loopback-mirror in the usual manner with a constant random offset.
Timestamp: The RTP timestamp denotes the sampling instant for when
the loopback-mirror is transmitting this packet to the loopback-
source. The RTP timestamp MUST be based on the same clock used by
the loopback-source. The initial value of the timestamp SHOULD be
random for security reasons (see Section 5.1 of RFC 3550
[RFC3550]).
SSRC: set as described in RFC 3550 [RFC3550].
CC and CSRC fields are used as described in RFC 3550 [RFC3550].
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7.1.2 RTP Payload Structure
The RTP header in the encapsulated packet MUST be followed by the
payload header defined in this section. If the received RTP packet
has to be looped back in multiple packets due to fragmentation, the
RTP header in each packet MUST be followed by the payload header
defined in this section. The header is devised so that the
loopback-source can usefully decode looped back packets in the
presence of moderate packet loss [RFC3550].
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| receive timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| F | R | CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| transmit timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The 12 octets after the receive timestamp are identical to the RTP
header in the received packet except for the first 4 bits of the
first octet.
Receive Timestamp: 32 bits
The Receieve timestamp denotes the sampling instant for when the
last octet of the media packet that is being encapsulated by the
loopback-mirror is received from the loopback-source. The Receive
timestamp MUST be based on the same clock used by the loopback-
source. The initial value of the timestamp SHOULD be random for
security reasons (see Section 5.1 of RFC 3550 [RFC3550]).
Fragmentation (F): 2 bits
First Fragment (00) /Last Fragment (01) /No Fragmentation(10)/
Intermediate Fragment (11). This field identifies how much of the
received packet is encapsulated in this packet by the loopback-
mirror. If the received packet is not fragmented, this field is
set to 10; otherwise the packet that contains the first fragments
sets this field to 00, the packet that contains the last fragment
sets this field to 01, all other packets set this field to 11.
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Reserved: 2 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
Any padding octets in the original packet MUST not be included in
the loopback packet generated by a loopback-mirror. The
loopback-mirror MAY add padding octets if required.
7.1.3 Usage of SDP
The payload type number for the encapsulated stream can be
negotiated using a mechanism like SDP. There is no static payload
type assignment for the encapsulated stream, so dynamic payload
type numbers MUST be used. The binding to the name is indicated by
an rtpmap attribute. The name used in this binding is "encaprtp".
The following is an example SDP fragment for encapsulated RTP.
m=audio 41352 RTP/AVP 112
a=rtpmap:112 encaprtp/8000
7.2 Direct loopback RTP payload format
The direct loopback RTP payload format can be used in scenarios
where the 16 byte overhead of the encapsulated payload format is
significant. This payload format MUST not be used in cases where
the MTU on the loopback path is less than the MTU on the transmit
path. When using this payload format, the receiver MUST loop back
each received packet in a separate RTP packet.
7.2.1 Usage of RTP Header fields
Payload Type (PT): The assignment of an RTP payload type for this
packet format is outside the scope of this document; it is either
specified by the RTP profile under which this payload format is
used or more likely signaled dynamically out-of-band (e.g., using
SDP; section 7.2.3 defines the name binding).
Marker (M) bit: Set to the value in the received packet.
Extension (X) bit: Defined by the RTP Profile used.
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Sequence Number: The RTP sequence number SHOULD be generated by the
loopback-mirror in the usual manner with a constant random offset.
Timestamp: The RTP timestamp denotes the sampling instant for when
the loopback-mirror is transmitting this packet to the
loopback-source. The RTP timestamp MUST be based on the same clock
used by the loopback-source. The initial value of the timestamp
SHOULD be random for security reasons (see Section 5.1 of RFC 3550
[RFC3550]).
SSRC: set as described in RFC 3550 [RFC3550].
CC and CSRC fields are used as described in RFC 3550 [RFC3550].
7.2.2 RTP Payload Structure
This payload format does not define any payload specific headers.
The loopback-mirror simply copies the payload data from the payload
portion of the packet received from the loopback-source.
7.2.3 Usage of SDP
The payload type number for the payload loopback stream can be
negotiated using a mechanism like SDP. There is no static payload
type assignment for the stream, so dynamic payload type numbers
MUST be used. The binding to the name is indicated by an rtpmap
attribute. The name used in this binding is "rtploopback".
The following is an example SDP fragment for encapsulated RTP.
m=audio 41352 RTP/AVP 112
a=rtpmap:112 rtploopback/8000
8. RTCP Requirements
The use of the loopback attribute is intended for monitoring of
media quality of the session. Consequently the media performance
information should be exchanged between the offering and the
answering entities. An offering or answering entity that is
compliant to this specification SHOULD support RTCP per [RFC3550]
and RTCP-XR per RFC 3611 [RFC3611]. Furthermore, if the client or
the server choose to support RTCP-XR, they SHOULD support RTCP-XR
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Loss RLE report block, Duplicate RLE report block, Statistics
Summary report block, and VoIP Metric Reports Block per sections
4.1, 4.2, 4.6, and 4.7 of RFC 3611 [RFC3611]. The client and the
server MAY support other RTCP-XR reporting blocks as defined by RFC
3611 [RFC3611].
9. Congestion Control
All the participants in a loopback session SHOULD implement
congestion control mechanisms as defined by the RTP profile under
which the loopback mechanism is implemented. For audio video
profiles, implementations SHOULD conform to the mechanism defined
in Section 2 of RFC 3551.
10. Examples
This section provides examples for media descriptions using SDP for
different scenarios. The examples are given for SIP-based
transactions and are abbreviated and do not show the complete
signaling for convenience.
10.1 Offer for specific media loopback type
A client sends an INVITE request with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49170 RTP/AVP 0
a=loopback:rtp-media-loopback
a=loopback-source:0
The client is offering to source the media and expects the server
to mirror the RTP stream per rtp-media-loopback loopback type.
A server sends a response with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
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i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49270 RTP/AVP 0
a=loopback:rtp-media-loopback
a=loopback-mirror:0
The server is accepting to mirror the media from the client at the
media level.
10.2 Offer for choice of media loopback type
A client sends an INVITE request with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49170 RTP/AVP 0 112 113
a=loopback:rtp-media-loopback rtp-pkt-loopback
a=loopback-source:0
a=rtpmap:112 encaprtp/8000
a=rtpmap:113 rtploopback/8000
The client is offering to source the media and expects the server
to mirror the RTP stream at either the media or rtp level.
A server sends a response with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49270 RTP/AVP 112
a=loopback:rtp-pkt-loopback
a=loopback-mirror:0
a=rtpmap:112 encaprtp/8000
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The server is accepting to mirror the media from the client at the
packet level using the encapsulated RTP payload format.
10.3 Offer for choice of media loopback type with rtp-start-loopback
A client sends an INVITE request with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49170 RTP/AVP 0 112 113
a=loopback:rtp-media-loopback rtp-pkt-loopback
a=loopback-source:0
a=rtpmap:112 encaprtp/8000
a=rtpmap:113 rtploopback/8000
m=audio 49170 RTP/AVP 100
a=loopback:rtp-start-loopback
The client is offering to source the media and expects the server
to mirror the RTP stream at either the media or rtp level. The
client also expects the server to source media until it receives
packets from the server per media described with the
rtp-start-loopback attribute.
A server sends a response with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49270 RTP/AVP 113
a=loopback:rtp-pkt-loopback
a=loopback-mirror:0
a=rtpmap:113 rtploopback/8000
m=audio 49270 RTP/AVP 100
a=rtpmap:100 pcmu/8000
a=loopback:rtp-start-loopback
The server is accepting to mirror the media from the client at the
packet level using the direct loopback RTP payload format. The
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server is also accepting to source media until it receives media
packets from the client.
10.4 Response to INVITE request rejecting loopback media
A client sends an INVITE request with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49170 RTP/AVP 0
a=loopback:rtp-media-loopback
a=loopback-source:0
The client is offering to source the media and expects the server
to mirror the RTP stream at the media level.
A server sends a response with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 0 RTP/AVP 0
a=loopback:rtp-media-loopback
a=loopback-mirror:0
NOTE: Loopback request may be rejected by either not including the
loopback mode attribute (for backward compatibility) or setting the
media port number to zero, or both, in the response.
10.5 Response to INVITE request rejecting loopback media with
rtp-start-loopback
A client sends an INVITE request with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
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s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 49170 RTP/AVP 0
a=loopback:rtp-media-loopback
a=loopback-source:0
m=audio 49170 RTP/AVP 100
a=loopback:rtp-start-loopback
The client is offering to source the media and expects the server
to mirror the RTP stream at the media level. The client also
expects the server to source media until it receives packets from
the server per media described with the rtp-start-loopback
attribute.
A server sends a response with SDP which looks like:
v=0
o=user1 2890844526 2890842807 IN IP4 192.0.2.11
s=Example
i=An example session
e=user@example.com
c=IN IP4 192.0.2.12/127
t=0 0
m=audio 0 RTP/AVP 0
a=loopback:rtp-media-loopback
a=loopback-mirror:0
m=audio 0 RTP/AVP 0
a=loopback:rtp-start-loopback
NOTE: Loopback request may be rejected by either not including the
loopback mode attribute (for backward compatibility) or setting the
media port number to zero, or both, in the response.
11. Security Considerations
The security considerations of [RFC3261] apply. Furthermore, given
that media loopback may be automated without the end user's
knowledge, the server of the media loopback should be aware of
denial of service attacks. It is recommended that sessions with
media loopback are authenticated and the frequency of such sessions
is limited by the server.
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12. Implementation Considerations
The media loopback approach described in this document is a
complete solution that would work under all scenarios. However, it
is believed that the solution may not be light-weight enough for
the common case. In light of this concern, this section clarifies
which features of the loopback proposal MUST be implemented for all
implementations and which features MAY be deferred if the complete
solution is not desired.
All implementations MUST support the rtp-pkt-loopback option for
loopback-type attribute. In addition, for the loopback-mode
attribute, all implementations of an offerer MUST at a minimum be
able to act as a loopback-source. All implementation MUST also at a
minimum support the direct media loopback payload type. Remaining
attribute values including rtp-media-loopback and
rtp-start-loopback MAY be implemented in complete implementations
of this draft.
13. IANA Considerations
13.1 SDP Attributes
This document defines three new media-level SDP attributes. IANA
has registered the following attributes:
Contact name: Kaynam Hedayat <khedayat@brixnet.com>.
Attribute name: "loopback".
Type of attribute: Media level.
Subject to charset: No.
Purpose of attribute: The 'loopback' attribute is used to
indicate the type of media loopback.
Allowed attribute values: The parameters to 'loopback' may be
one or more of "rtp-pkt-loopback,"
"rtp-media-loopback," and
"rtp-start-loopback". See section 5
of this document for syntax.
Contact name: Kaynam Hedayat <khedayat@brixnet.com>.
Attribute name: "loopback-source".
Type of attribute: Media level.
Subject to charset: No.
Purpose of attribute: The 'loopback-source' attribute
specifies that the sender is the media
source and expects the receiver to act
as a loopback-mirror.
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Allowed attribute values: The parameter to 'loopback-source' is
a media format ("<fmt>") description
as defined in RFC 4566 Section 5.14.
Contact name: Kaynam Hedayat <khedayat@brixnet.com>.
Attribute name: "loopback-mirror".
Type of attribute: Media level.
Subject to charset: No.
Purpose of attribute: The 'loopback-mirror' attribute
specifies that the receiver will
mirror (echo) all received media back
to the sender of the RTP stream.
Allowed attribute values: The parameter to 'loopback-mirror' is
a media format ("<fmt>") description
as defined in RFC 4566 Section 5.14.
13.2 MIME Types
The IANA has registered the following MIME types:
13.2.1 audio/encaprtp
To: ietf-types@iana.org
Subject: Registration of media type audio/encaprtp
Type name: audio
Subtype name: encaprtp
Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
Optional parameters: none
Encoding considerations: This media type is framed
binary data.
Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
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within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
group delegated from the IESG.
13.2.2 video/encaprtp
To: ietf-types@iana.org
Subject: Registration of media type video/encaprtp
Type name: video
Subtype name: encaprtp
Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
Optional parameters: none
Encoding considerations: This media type is framed
binary data.
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Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
group delegated from the IESG.
13.2.3 text/encaprtp
To: ietf-types@iana.org
Subject: Registration of media type text/encaprtp
Type name: text
Subtype name: encaprtp
Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
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Optional parameters: none
Encoding considerations: This media type is framed
binary data.
Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
group delegated from the IESG.
13.2.4 application/encaprtp
To: ietf-types@iana.org
Subject: Registration of media type
application/encaprtp
Type name: application
Subtype name: encaprtp
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Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
Optional parameters: none
Encoding considerations: This media type is framed
binary data.
Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
group delegated from the IESG.
13.2.5 audio/rtploopback
To: ietf-types@iana.org
Subject: Registration of media type audio/rtploopback
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Type name: audio
Subtype name: rtploopback
Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
Optional parameters: none
Encoding considerations: This media type is framed
binary data.
Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
group delegated from the IESG.
13.2.6 video/rtploopback
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To: ietf-types@iana.org
Subject: Registration of media type video/rtploopback
Type name: video
Subtype name: rtploopback
Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
Optional parameters: none
Encoding considerations: This media type is framed
binary data.
Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
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group delegated from the IESG.
13.2.7 text/rtploopback
To: ietf-types@iana.org
Subject: Registration of media type text/rtploopback
Type name: text
Subtype name: rtploopback
Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
Optional parameters: none
Encoding considerations: This media type is framed
binary data.
Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
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Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
group delegated from the IESG.
13.2.8 application/rtploopback
To: ietf-types@iana.org
Subject: Registration of media type
application/rtploopback
Type name: application
Subtype name: rtploopback
Required parameters:
rate:RTP timestamp clock rate, which is equal to the
sampling rate. The typical rate is 8000; other rates
may be specified.
Optional parameters: none
Encoding considerations: This media type is framed
binary data.
Security considerations: See Section 11 of this document.
Interoperability considerations: none
Published specification: This MIME type is described fully
within this document.
Applications which use this media type: Applications wishing
to monitor and ensure the quality of transport to the
edge of a given VoIP, Real-Time Text or Video Over IP
Service.
Additional information: none
Person & email address to contact for further information:
Kaynam Hedayat
EMail: khedayat@brixnet.com
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Intended usage: COMMON
Restrictions on usage: This media type depends on RTP
framing, and hence is only defined for transfer via
RTP. Transfer within other framing protocols is not
defined at this time.
Author:
Kaynam Hedayat.
Change controller: IETF Audio/Video Transport working
group delegated from the IESG.
14. Acknowledgements
The authors wish to thank Nagarjuna Venna, Flemming Andreasen, Jeff
Bernstein, Paul Kyzivat, and Dave Oran for their comments and
suggestions.
15. Normative References
[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.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
Model with the Session Description Protocol (SDP)",
RFC 3264, June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R. and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3611] Almeroth, K., Caceres, R., Clark, A., Cole, R.,
Duffield, N., Friedman, T., Hedayat, K., Sarac, K.
and M. Westerlund, "RTP Control Protocol Extended
Reports (RTCP XR)", RFC 3611, November 2003.
[RFC4234] Crocker, P. Overell, "Augmented ABNF for Syntax
Specification: ABNF", RFC 4234, October 2005.
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[RFC2119] Bradner, S.,"Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2736] Handley, M., Perkins, C., "Guidelines for Writers of
RTP Payload Format Specifications", RFC 2736, BCP
0036, December 1999.
[RFC3551] Schulzrinne, H., Casner, S., "RTP Profile for Audio
and Video Conferences with Minimial Control", STD 65,
RFC 3551, July 2003.
[RFC4566] Handley, M., Jacobson, V., Perkins, C., "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
Authors' Addresses
Kaynam Hedayat
Brix Networks
285 Mill Road
Chelmsford, MA 01824
US
Phone: +1 978 367 5611
EMail: khedayat@brixnet.com
URI: http://www.brixnet.com/
Nagarjuna Venna
Brix Networks
285 Mill Road
Chelmsford, MA 01824
US
Phone: +1 978 367 5703
EMail: nvenna@brixnet.com
URI: http://www.brixnet.com/
Paul E. Jones
Cisco Systems, Inc.
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7025 Kit Creek Rd.
Research Triangle Park, NC 27709
US
Phone: +1 919 392 6948
EMail: paulej@packetizer.com
URI: http://www.cisco.com/
Arjun Roychowdhury
Hughes Systique Corp.
15245 Shady Grove Rd, Ste 330
Rockville MD 20850
US
Phone: +1 301 527 1629
EMail: arjun@hsc.com
URI: http://www. hsc.com/
Chelliah SivaChelvan
Cisco Systems, Inc.
2200 East President George Bush Turnpike
Richardson, TX 75082
US
Phone: +1 972 813 5224
EMail: chelliah@cisco.com
URI: http://www.cisco.com/
Nathan Stratton
BlinkMind, Inc.
2027 Briarchester Dr.
Katy, TX 77450
Phone: +1 832 330 3810
EMail: nathan@robotics.net
URI: http://www.robotics.net/
Full Copyright Statement
Copyright (C) The IETF Trust (2008).
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This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.
This document and the information contained herein are provided on
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REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE
IETF TRUST AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL
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Acknowledgement
Funding for the RFC Editor function is provided by the IETF
Administrative Support Activity (IASA).
Hedayat, et al. Expires January 2009 [Page 33]