One document matched: draft-hancock-sip-interconnect-guidelines-02.txt
Differences from draft-hancock-sip-interconnect-guidelines-01.txt
Speermint D. Hancock
Internet-Draft D. Malas
Intended status: Informational CableLabs
Expires: April 29, 2010 October 26, 2009
draft-hancock-sip-interconnect-guidelines-02
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Abstract
As Session Initiation Protocol (SIP) peering becomes more widely
accepted by service providers the need to define an interconnect
guideline becomes of greater value. This document takes into
consideration the SIP and commonly used SIP extensions, and it
defines a fundamental set of requirements for SIP Service Providers
(SSPs) to implement within their signaling functions (SFs) or
Signaling Path Border Elements (SBEs) for peering.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
1.1. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 7
2.1. Requirements Language . . . . . . . . . . . . . . . . . . 7
3. Reference Architecture . . . . . . . . . . . . . . . . . . . . 8
4. General Procedures . . . . . . . . . . . . . . . . . . . . . . 10
4.1. Extension Negotiation . . . . . . . . . . . . . . . . . . 10
4.2. Public User Identities . . . . . . . . . . . . . . . . . . 10
4.2.1. Identifying the Called User . . . . . . . . . . . . . 10
4.2.2. Identifying the Calling User . . . . . . . . . . . . . 11
4.3. Trust Domain and Asserted Identities . . . . . . . . . . . 11
4.4. IPv4/6 Interworking . . . . . . . . . . . . . . . . . . . 11
4.5. Fault Isolation and Recovery . . . . . . . . . . . . . . . 11
4.5.1. Interface Failure Detection . . . . . . . . . . . . . 11
4.5.2. Overload Control . . . . . . . . . . . . . . . . . . . 12
4.5.3. Session Timer . . . . . . . . . . . . . . . . . . . . 12
5. Call Features . . . . . . . . . . . . . . . . . . . . . . . . 13
5.1. Session Establishment . . . . . . . . . . . . . . . . . . 13
5.1.1. SDP Requirements . . . . . . . . . . . . . . . . . . . 13
5.1.2. Basic Call Setup . . . . . . . . . . . . . . . . . . . 13
5.1.3. Ringback Tone vs. Early Media . . . . . . . . . . . . 14
5.1.4. Early-Media with Multiple Terminating Endpoints . . . 15
5.1.5. Establishing calls using 3PCC . . . . . . . . . . . . 16
5.1.6. Hold . . . . . . . . . . . . . . . . . . . . . . . . . 16
5.2. Calling Name and Number Deliver (with Privacy) . . . . . . 17
5.3. Call Forwarding . . . . . . . . . . . . . . . . . . . . . 17
5.3.1. Detecting Call Forwarding Loops . . . . . . . . . . . 18
5.4. Call Transfer . . . . . . . . . . . . . . . . . . . . . . 19
5.4.1. Call-Transfer Using REFER/Replaces . . . . . . . . . . 19
5.4.2. Call-Transfer Using 3PCC . . . . . . . . . . . . . . . 20
5.5. 3-way Conference . . . . . . . . . . . . . . . . . . . . . 20
5.6. Auto Recall/Callback . . . . . . . . . . . . . . . . . . . 21
5.6.1. Originating SBE Sends INVITE to Target . . . . . . . . 21
5.6.2. Originating SBE Sends SUBSCRIBE to Target . . . . . . 21
5.6.3. Target Sends NOTIFY to Originating SBE . . . . . . . . 22
6. Security Considerations . . . . . . . . . . . . . . . . . . . 23
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 24
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 25
9. Normative References . . . . . . . . . . . . . . . . . . . . . 26
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 28
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1. Introduction
In the SIP Service Provider (SSP) industry every SSP has their own
SIP requirements. Whether they defined it themselves or a vendor's
equipment capabilities defined it for them, they have one. When two
SSPs approach one another to establish a peering relationship, one of
the first pieces of information they exchange is their respective SIP
requirements or profiles. (For the purposes of this draft, we will
call it a SIP profile.) After exchanging SIP profiles, each SSP will
likely go back to their lab and spend an extended period of time
attempting to comply with the other SSP's SIP profile, either by
requesting their vendor to implement or change capabilities, by
developing "interworking profiles" for manipulating SIP messages at
their SF or SBE, or by arguing defiantly that their approach is
correct. While this may seem like a simple and manageable task when
establishing a single peering relationsip, it can become extremely
burdensome as the number of peering relationships increase to four,
five, and beyond.
The overwhelming sentiment is that there is a need to establish a
minimum set of requirements an SSP can implement within their SF or
SBE to peer with any other SSP. While this may seem like an arduous
task, there is a belief that a fundamental set of requirements could
be established as a baseline guideline to establish peering with any
SSP. After the peering is established, the two SSPs may agree on
additional SIP parameters or extensions that expand the capabilities
for many different purposes. Over time, this document may be
extended or updated as necessary to maintain consistency with the
widely adopted new use of SIP functionality in the industry.
This document provides an interconnect guideline to address potential
SIP interworking issues for peering SIP-based networks.
1.1. Scope
This document describes the SIP interconnect procedures between two
SIP-based networks for both peering SIP Service Providers networks
and peering Enterprise networks.
The document focuses on the interworking procedures required to
support basic telephone service, including the following
capabilities:
o On-net to on-net calls
o Multi-media
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* Voice
* Video
* Real-time text
o Caller ID with Privacy
o Early media
o Local Number Portability
o Call hold/conf/xfer
o Call forwarding
o Auto Recall/Callback
o Problem Isolation - Inter-network keep-alives
Interworking procedures in support of the following capabilities are
not addressed:
o Calls to/from PSTN
o Operator calls
o 0+,0-, busy-line-verify
o Emergency calls
o Transmission loss plan
o Operational capabilities
o Accounting
o Electronic Surveillance
o Quality-of-Service
o Authentication and Security
o Voice, FAX, DTMF-relay
o RTCP VoIP Metrics
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o SIP RTP Loopback
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2. Terminology
2.1. Requirements Language
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 [RFC2119].
This draft also uses terms defined in [RFC5486].
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3. Reference Architecture
Figure 1 shows the peering relationship between two SSPs; SSP-A and
SSP-B. The Signaling Path Border Element (SBE) serves as the egress/
ingress point for SIP signaling into each peers network. The SBE may
act as a proxy or a Back-to-Back User Agent (B2BUA). The optional
Data Path Border Element (DBE) serves as a media relay at the peering
interface for media interworking, topology hiding and IPv4/6
interworking.
+------+
| DNS, |
+---------->| Db, |<---------+
| | etc | |
| +------+ |
| |
------|-------- -------|-------
/ v \ / v \
| +--LUF-+ | | +--LUF-+ |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| +------+ | | +------+ |
| | | |
| +--LRF-+ | | +--LRF-+ |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| +------+ | | +------+ |
| | | |
| | | |
| +---SF--+ +---SF--+ |
| | | | | |
| | SBE | | SBE | |
| Originating | | | | Target |
| +---SF--+ +---SF--+ |
| SSP | | SSP |
| +---MF--+ +---MF--+ |
| | | | | |
| | DBE | | DBE | |
| | | | | |
| +---MF--+ +---MF--+ |
\ / \ /
--------------- ---------------
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Figure 1: Peering Architecture
This document places requirements on the following two entities in
the Peering architecture:
o the SBE in support of the SIP/SDP interface between two peering
networks
o the DBE in support of the RTP/RTCP interface between peering
networks
Based on the definition of the SBE in the peering architecure
document [I-D.ietf-speermint-architecture], it's not entirely correct
to make the SBE responsible for meeting all the SIP signaling
requirements at the peering interface. For example, the SBE can play
the role of a firewall or a SIP proxy that is largely transparent to
the SIP messages crossing the interface. In reality, the
responsiblity for supporting this interface belongs to all the SIP
entities in the SIP signaling chain, including the SBE plus the SIP
proxies and UAs inside the network. Furthermore, when this network
is serving as a transit network, the SIP signaling chain can extend
into another network. To resolve this issue, this document extends
the definition of the SBE slightly so that when it is specified as
the target entity of a normative requirement on the SIP/SDP peering
interface, the SBE represents all the SIP entities in the SIP
signaling chain that can effect the SIP/SDP peering interface.
Likewise, the DBE can act as a media relay (performing no media
encode/decode) that is transparent to the RTP/RTCP packets exchanged
with the peer network. Therefore, this document extends the
definition of the DBE so that when it is specified as the target
entity of a normative requirement on the RTP/RTCP interface, it
represents the media endpoint in the peering network that terminates
the RTP/RTCP stream.
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4. General Procedures
4.1. Extension Negotiation
It is RECOMMENDED that originating SBEs facing other peer networks be
configured in such a way that they do not require any SIP extensions
to be supported by the other end. The SBE MUST identify all
supported SIP extensions in the Supported header. The SBE MAY
support configuration controls to disable certain extensions based on
bilateral agreement between peer networks. For example, the SBE
could be configured to remove '100rel' from the Supported header in
order to disable the use of reliable provisionable response (PRACK).
When sending a dialog-initialing request to a peer network, the SBE
MUST identify all supported SIP requests in the Allow header field.
4.2. Public User Identities
Users are identified at the peering interface by their Public User
Identity. An SBE MUST encode Public User Identities as a SIP URI of
the telephone-subscriber syntax form of a Tel URI as indicated by the
"user=phone" parameter (see Section 19.1.6 of [RFC3261]), where the
user part of the SIP URI contains a global Tel URI as defined in
[RFC3966].
Example:
sip:+13035551212@examplessp.com;user=phone
4.2.1. Identifying the Called User
When sending a dialog-initiating request to a peer network, the SBE
MUST :
o identify the called user in the Request-URI of the request, using
the telephone-subscriber syntax form of the SIP URI as described
above in section 4.2; and
o if Local Number Portability (LNP) information for the called
number was obtained, then
* include the LNP data in SIP URI in the Request-URI using the
Tel URI "npdi" and "rn" parameters as defined in [RFC4694], and
* if the called number is ported, then identify the routing
number using the global form of the "rn" parameter, which is
indicated by a leading "+" character followed by the country-
code followed by the national number (e.g., "rn=+16132220000").
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On receiving a dialog-initiating request from a peer network, the SBE
MUST:
o identify the called user based on the contents in the Request-URI,
where the Request URI contains a SIP URI as described above in
section 4.2, and
o obtain the LNP data for the called number based on the presence
and contents of the "npdi" and "rn" parameters contained in the
SIP URI of the Request-URI as defined in [RFC4694].
4.2.2. Identifying the Calling User
When sending or receiving a dialog-initiating request, the SBE MUST:
o identify the calling user in the P-Asserted-Identity header using
the telephone-subscriber syntax form of the SIP URI as described
above in section 4.2; and
o if calling name display is supported, then include the calling
name display information in the P-Asserted-Identity header as
described in section 5.2.
4.3. Trust Domain and Asserted Identities
In a peering relationship, both originating and terminating networks
are in the same trust domain. Therefore, per [RFC3325], the
terminating network MUST trust an originating peer network to
populate the P-Asserted-Identity header in an incoming INVITE request
with the Public User Identity of the originating user. Furthermore,
the originating network MUST trust the terminating network to honor
the privacy wishes of the originator as indicated in the Privacy
header.
4.4. IPv4/6 Interworking
It is the responsibility of the IPv6 network to perform the IPv4/IPv6
interworking function when interworking with an IPv4 network.
4.5. Fault Isolation and Recovery
4.5.1. Interface Failure Detection
An SBE MAY periodically send an OPTIONS request with Max-forwards set
to '0' to detect the availability of a peer's ingress point. The
ping rate is based on bi-lateral agreement (typically every 5
seconds). If the sending SBE fails to receive a response to an
OPTIONS request, then it will consider that non-responding ingress
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point into the peer network to have failed, and will refrain from
routing new requests to it. In the meantime, it will continue to
send periodic OPTIONS pings to the failed ingress point in order to
detect when it has been restored and is available for service.
Note: A possible enhancement to the OPTIONS ping is to declare a
well-known SIP URI in the look-up function (LUF) that could be used
to test the health of each ingress SF or SBE in a peer network. For
example, SIP INVITE (with no SDP) to sip:999999999@examplessp.com
would respond with a 200 OK (again no SDP), followed by a BYE/200 OK.
4.5.2. Overload Control
SIP does not currently provide an explicit overload control
mechanism. However, an SSP may want to impose limits on the number
of simultaneous calls, and the incoming call rate it will accept from
a peer SSP. On receiving a dialog-initiating request that exceeds
such limits, the receiving SBE MUST respond with a 503 (Service
Unavailable) response. An SBE receiving a dialog-initiating request
from a peer MUST limit the use of the 503 (Service Unavailable)
response to reporting overload at the ingress signaling point, and
MUST NOT use this response to report overload or other failures
internal to the network.
On receiving a 503 (Service Unavailable) response from a peer
network, the receiving SBE MUST limit the scope of the response to
the call on which it was received (i.e., a 503 response has no affect
on the routing of subsequent calls to the peer). Also, the receiving
SBE MUST attempt to consume the 503 (Service Unavailable) response
from a peer as close to the egress signaling point as possible, and
avoid propagating the response back toward the source of the request.
Specifically, on receiving a 503 (Service Unavailable) response to a
dialog-initiating request that was sent to a peer network, the SBE
MUST:
o terminate the current transaction,
o ignore the Retry-After header if one is present, and
o attempt to route the call via an alternate peering interface (i.e.
do not attempt to route the call via the same peering interface
since it may encouter and aggravate the same overload condition).
4.5.3. Session Timer
The SBE SHOULD support Session Timer as defined in [RFC4028].
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5. Call Features
5.1. Session Establishment
5.1.1. SDP Requirements
The SBE MUST support the SDP requirements defined in [RFC4566]. The
SBE MUST include only one media (m=) descriptor per desired media
stream in an SDP offer to a peer network.
If an SBE receives an SDP offer containing multiple media
descriptors, it MUST act on the media descriptors and include all of
them in the same order in the response, including non-zero ports and
zero ports for the offered media according to its capabilities as
specified in [RFC3264] An Offer/Answer Model with SDP. The SBE MUST
NOT reject an offered session because it offers more media than the
SBE can handle.
5.1.2. Basic Call Setup
This section describes the procedures at the peering interface
required to establish a 2-way session for a basic voice call between
two users. In this case it is assumed that no originating or
terminating features are applied (no call blocking, forwarding,
etc.), and that the called line is available to accept the call.
Also, this section describes the session establishment procedures
when tha call is initiated by the originating SIP User Agent itself,
and not via a 3rd party in support of features like click-to-call.
Two-way call establishment using 3PCC is covered in section
Section 5.1.5.
The SBE MUST support the SDP offer/answer procedures specified in
[RFC3264]. The originating SBE MUST include an SDP offer in the
initial INVITE. The terminating SBE MUST include an SDP answer in
the first reliable response to INVITE (typically the final 200 (OK)
response to the INVITE). The terminating SBE MAY also include an SDP
body in a non-reliable provisional 18x response to the INVITE. The
SDP contained in a non-reliable 18x provisional response can be
considered a "preview" of the actual SDP answer to be sent in the 200
(OK) to INVITE. The originating SBE can act on this "preview" SDP to
establish an early media session, as described in section
Section 5.1.3. The terminating SBE MUST ensure that the "preview"
SDP matches the actual SDP answer contained in the 200 (OK) response
to INVITE.
Note: an SDP offer/answer exchange occurs within the context of a
single dialog. Therefore, the requirement for matching SDPs in the
provisional and final responses to INVITE applies only when the
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provisional and final response are in the same dialog. If the
provisional and final response are on different dialogs (say, when
the INVITE is forked), the requirement for matching SDPs does not
apply.
The SBE MUST always set the SDP mode attribute in the initial offer/
answer to "a=sendrecv".
Note: Setting the mode to "a=sendrecv" on the initial SDP offer/
answer exchange avoids an additional SDP offer/answer exchange to
update the mode to send-receive after the call is answered. This
should help mitigate the problem of voice-clipping on answer.
A peered originating and terminating SBE that advertise support for
different but overlapping sets of codecs in the SDP offer/answer
exchange for a given call MUST negotiate a common codec for the call.
5.1.3. Ringback Tone vs. Early Media
During the call setup phase, while the originating network is waiting
for the terminating network to answer the call, the originating line
is either playing local ringback tone to the calling user or
connected to a receive-only or bi-directional early-media session
with the terminating network. For example, early media can be
supplied by the terminating endpoint (e.g., custom ringback tone)
while waiting for answer.
During session establishment, an SBE MUST use the following
procedures to control whether the originating line applies local
ringback tone or establishes an early media session while waiting for
the call to be answered:
o The terminaing SBE MUST send the following provisional response to
a call-initiating INVITE:
* a 180 (Alerting) response containing no SDP if the call
scenario requires the originating network to apply local
ringback tone,
* a 183 (Progressing) response containing SDP that describes the
terminating media endpoint if the call scenario requires the
originating network to establish an early-media session with
the terminating media endpoint,
* the provisional response sent for other call scenarios is not
specified, as long as the response is not one of those
specified above.
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o The originating SBE MUST perform the following action on receipt
of a provisional response to a call-initiating INVITE:
* on receiving a 180 (Alerting) response containing no SDP, apply
local ringback tone,
* on receiving a 180 (Alerting) or 183 (Progressing) response
containing SDP, establish an early media session with the media
endpoint described by the SDP,
* on receiving any other provisional response (with or without
SDP) do nothing (e.g., continue to apply local ringback tone if
it was already being applied when response was received)
5.1.4. Early-Media with Multiple Terminating Endpoints
There are some call scenarios that require media sessions to be
established (serially) between the originating user agent and one or
more intermediate media endpoints before the call is connected to the
final target called user agent. For example, the terminating network
can insert a media server in the call to interact with the calling
user in some way (e.g., to collect a blocking-override PIN) before
offering the call to the called user. Another case occurs when the
called user fails to answer within an allotted time and the call is
redirected to voice-mail, or forwarded to another user via Call
Forwarding Don't Answer (CFDA). These different cases can be
combined in the same call.
For each terminating media endpoint that is associated with a call
before the call is answered, the terminating network must decide
whether to establish an early media session or apply ringback tone at
the originating user agent. For example, consider the case where the
called user has call blocking with PIN override, and CFDA. First, an
early-media session is established with the call-blocking server to
collect the PIN, next the originating user agent is instructed to
play local ring-back tone while waiting for the called user to
answer, and finally an early media session is established with the
forward-to party to play custom ringback tone.
[RFC3261] mandates that the SDP included unreliable provisional 18x
responses to INVITE within the context of a dialog must match the
SDP-answer included in the final 200 (OK) response to INVITE. The
following sections describe two different mechanisms for supporting
multiple terminating media endpoints before answer, within the
confines of this requirement.
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5.1.4.1. Forking the INVITE
For each terminating media endpoint that requires an early media
session to be established with the originating media endpoint, the
terminating SBE MUST signal the attributes of the terminating media
endpoint to the originating SBE within the SDP of a 183 (Progressing)
response. The terminating SBE MUST ensure that 18x responses
containing different SDP copies are not sent within the same dialog.
The terminating SBE does this by specifying a different To: tag for
each provisional response that contains a unique SDP, as if the
INVITE had been sequentially forked.
The originating SBE MUST honor the most recently received 18x
response to INVITE, based on the procedures defined in section
5.1.2.2.
5.1.4.2. Redirecting the INVITE
As an alternative to sequentially forking the INVITE, the terminating
entity can redirect the originating entity to the next endpoint in
the series by sending a 302 (Moved Temporarily) response containing a
Contact header field that identifies the next endpoint. The
resulting INVITE from the originating SBE is sent as a dialog-
initiating request, and can therefore establish a new early-media
session with the next endpoint in the series. The use of this
procedure is based on bilateral agreement between peering operators.
On receiving a 302 (Moved Temporarily) response to an INVITE request,
and if this mechanism is enabled based on local policy, the
originating SBE MUST send a new dialog-initiating INVITE with a
Request-URI set to the value returned in the Contact header field of
the 302 (Moved Temporarily) response, as described in [RFC3261].
5.1.5. Establishing calls using 3PCC
Section Section 5.1.2 describes the procedures that are used to
establish basic two-way call when the call is initiated directly by
the originating user's endpoint. However, an SSP may support
features such as click-to-call, where the call is initiated by a 3rd
party such as an Applicaion Server on bahalf of the originating user.
To support such features, the SBE MUST support the 3PCC procedures
described in [RFC3725].
5.1.6. Hold
An SBE that wishes to place a media stream "on hold" MUST offer an
updated SDP to its peer network with an attribute of "a=inactive" or
"a=sendonly" in the media description block. An SBE that wishes to
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place a media stream "on hold" MUST NOT set the connection
information of the SDP to a null IP address (for example, it MUST NOT
set the 'c=' connection line to c=IN IP4 0.0.0.0). An SBE that wants
to place a media stream "on hold" SHOULD locally mute the media
stream.
An SBE that receives an SDP offer with an attribute of "a=inactive"
in the media block MUST place the media stream "on hold", and MUST
answer with an updated SDP containing a media attribute of
"a=inactive". An SBE that receives an SDP offer with an attribute of
"a=inactive" in the media block MUST NOT set the connection data of
the answer SDP to c=0.0.0.0. An SBE operating in IPv4 that receives
an SDP offer with no directionality attributes but connection data
set to c=IN IP4 0.0.0.0 SHOULD place the media stream "on hold".
5.2. Calling Name and Number Deliver (with Privacy)
An originating SBE MUST provide the calling name and number of the
originating user in the P-Asserted-Identity header of dialog-
initiating requests. (The mechanism for obtaining the calling name
is out-of-scope of this document.) The calling number is contained
in the telephone-subscriber syntax form of the SIP URI, containing an
E.164 number as described in section 4.2. The calling name is
contained in the display-name component of the P-Asserted-Identity
header.
If the originating user wants to remain anonymous, the originating
SBE MUST include a Privacy header containing the value "id" as
specified in [RFC3323] and [RFC3325]. In addition, the originating
SBE SHOULD obscure the identity of the originating user in other
headers as follows:
o Set the identity information in the 'From' header to "Anonymous
sip:anonymous@anonymous.invalid",
o Set the display-name in the To header to "Anonymous" (since the To
display-name selected by the originating user could provide a hint
to the originating user's identity).
o Obscure any information from the Call-ID and Contact headers, such
as the originating FQDN, that could provide a hint to the
originating user's identity.
5.3. Call Forwarding
If an SSP offers call-forwarding services to its users, then the
forwarding SBE MAY remain in the signaling path of the forwarded call
in order to support separate billing of forward-from and forward-to
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legs. This is accomplished by
o remaining in the signaling path as a proxy or B2BUA, or
o by responding to the initial INVITE with a 302 (Moved Temporarily)
response with a Contact header containing a private URI that
points back to the forwarding network
MUST support the History-Info header as defined in [RFC4244] to
detect call-forwarding loops.
5.3.1. Detecting Call Forwarding Loops
A call forwarding loop is defined to be the scenario that occurs when
a targeted subscriber for a call forwards the call to another
destination. If the forwarded-to destination also has call
forwarding configured, the call can forward back (directly or
indirectly) to the original targeted subscriber. When a loop is
detected, the network that performs the detection rejects the call.
An SBE MUST detect call forwarding loops. An SBE MUST support a
configurable limit on the number of times an individual call may be
subject to forwarding. If the number of forwarding attempts for a
single call exceeds this limit, the SBE MUST reject the call.
An SBE SHOULD detect call forwarding loops and limit the number times
a call is forwarded by supporting the History-Info header field as
defined in [RFC4244], and by analysing the History-Info entries as
described in this section. However, these mechanisms alone may not
be sufficient to detect loops when calls are forwarded to networks
not supporting these mechanisms. Therefore an SBE MAY support
additional loop prevention and forwarding limit detection methods as
long as the requirements of forwarding limit restriction and loop
detection are met.
If the SBE supports the prevention of forwarding loops via analysis
of the History-Info header present in the INVITE then it MUST compare
the forward-to address with the set of targeted-to URI (hi-targeted-
to-uri) entries from the History-Info header. If there is a match
then a loop has occurred. If no History-Info header is present then
it is not possible to perform loop detection via this mechanism.
If an SBE supports the prevention of forwarding loops by enforcing a
maximum number of forwarding attempts, then it MUST calculate the
number of forwarding attempts by counting the number of entries in
the History-Info header that were added due to call forwarding (i.e.,
entries containing a nested Reason header which includes a protocol-
cause parameter and a reason-text parameter that indicate the call
was forwarded as defined below). If no History-Info header is
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present then it is not possible to determine the number of forwarding
attempts via this mechanism.
[RFC4458] defines the mapping between the forwarding conditions and
the coding of the protocol-cause parameter in the Reason header. An
SBE MUST populate the Reason header with a protocol-cause value of
"486" and a reason-text value of "CFBL" when the forwarding condition
is Call Forwarding Busy Line (CFBL). The SBE MUST populate the
Reason header with a protocol-cause value of "408" and a reason-text
value of "CFDA" when the forwarding condition is Call Forwarding
Don't Answer (CFDA). The SBE MUST populate the Reason header with a
protocol-cause value of "302" and a reason-text value of "CFV/SCF"
when the forwarding ondition is Caff Forwarding Valriable (CFV) or
Selective Call Forwarding (SCF).
5.4. Call Transfer
A user in a peered call can perform the various forms of call-
transfer (consultive transfer, blind transfer). Call-transfer can be
supported in one of two ways; either using the REFER request
[RFC3515] and Replaces header [RFC3891], or by manipulating the call
legs using 3rd Party Call Control (3PCC) techniques. SBEs that
support call transfer MUST support the 3PCC option, and MAY support
the REFER/Replaces option. If a network supports both options, then
the option that is used when interworking with a specific peer is
based on locally configured data that indicates the capabilities of
that peer.
5.4.1. Call-Transfer Using REFER/Replaces
SBEs that support call-transfer using the procedures described in
this section MUST support the SIP REFER extension described in
[RFC3515], and the SIP Replaces extension described in [RFC3891].
Furthermore, [RFC3515] requires support of the SIP Event Notification
extension described in [RFC3265].
To describe the basic transfer call-flow, consider the case where
user-A in SSP-A is in an active call with user-B in peered sSSP-B,
and user-A decides to transfer user-B to user-C. User-C could be
located anywhere in the global network; for example in SSP-A, SSP-B,
another peered network, a non-peering IP network, or the PSTN. Here
are the basic steps to complete the transfer using REFER/Replaces:
o User-A puts user-B on hold (sends re-INVITE with SDP "a=inactive"
as described in 5.1.1.1)
o User-A initiates a basic 2-way call to user-C
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o User-A sends an in-dialog REFER to user-B containing a Refer-To
header. The Refer-To header instructs user-B to send an INVITE to
user-C with an imbedded Replaces header identifying the A-to-C
dialog.
* If SSP-A is not required to remain in the signaling path of the
transferred call, then it identifies user-C directly in the
Refer-To header,
* If SSP-A is required to remain in the signaling path of the
transferred call (say to generate events for proper billing of
the call), then it identifies a private URL pointing to itself
in the Refer-To header. The private URL contains a "hostport"
that identifies SSP-A, and contains a "userinfo" string that is
generated by SSP-A. This "userinfo" string either contains or
points to information required by SSP-A to establish the
transfer-to leg of the call.
o User-B sends an INVITE containing the Replaces header specified in
step-3 to the address contained in the Refer-To header (i.e., the
INVITE is routed to user-C either directly from ssp-B, or
indirectly via SSP-A using the private URL)
o User-B sends Notify requests within the original A-to-B dialog,
informing user-A of the progress of the B-to-C call
o At some point user-A drops out of both dialogs (e.g., drops out of
A-to-C dialog on receiving BYE from user-C). At this point users
B and C are active in a 2-way call.
SBEs SHOULD support receiving a GRUU [I-D.ietf-sip-gruu] in the
Refer-To header.
5.4.2. Call-Transfer Using 3PCC
SBEs that support call-transfer using 3PCC techniques MUST act as a
B2BUA, and manipulate the call legs using INVITE and re-INVITE
requests. It is RECOMMENDED that such techniques follow the guidance
presented in [RFC3725].
5.5. 3-way Conference
The media mixing for 3-way conference calls may be performed by the
user agent of the conference control party, or by a conference bridge
application server in the peer network serving the conference control
party. When mixing is done by the user agent, there are no specific
requirements placed on the peering interface other than the support
of hold as described in section 5.1.1.1. When conference mixing is
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performed by a network-based server, users are added to the
conference using procedures similar to those described for call
transfer in section 5.4.
5.6. Auto Recall/Callback
When a user invokes AC or AR, and the user targeted by the recall/
callback feature belongs to a peer network, the originating SBE first
attempts to establish a basic 2-way call with the target user. If
the call completes normally (e.g., the target user answers) then the
feature is complete. If the terminating SBE responds with an
indication that the target user is busy, then the originating SBE
subscribes to the dialog-event package as defined in [RFC4235] of the
target user, as a mechanism to detect when the target user becomes
available. When the terminating SBE subsequently notifies the
originating SBE that the target user is available, the originating
SBE re-attempts to establish a 2-way call to the target user.
5.6.1. Originating SBE Sends INVITE to Target
When a user invokes an AR or AC call, the originating SBE MUST follow
the procedures given for a basic call as described in Section 5.1.2,
and attempt to establish a 2-way call with the target user. In
addition, the originating SBE MUST add a Call-Info header field to
the INVITE with a purpose of "answer_if_not_busy".
If the originating SBE receives a 200-OK response to INVITE, then the
AC/AR feature is considered complete, and the remainder of the call
is handled like a normal 2-way call. If the originating SBE receives
a 486-Busy-Here or 600-Busy-Everywhere response to the INVITE, then
it MUST follow the AC/AR procedures as defined below. If the
terminating SBE receives an inbound INVITE with a Call-Info header
field declaring purpose= answer_if_not_busy, then the terminating SBE
MUST ignore any active Call-Forwarding-Busy-Line (CFBL) service for
the target user, not forward the call if the target is busy, and
instead handle the call as if CFBL was not active (e.g., offer the
call using the call-waiting feature).
5.6.2. Originating SBE Sends SUBSCRIBE to Target
On receiving a 486-Busy-Here or 600-Busy-Everywhere response to an
AC/AR INVITE request, the originating SBE MUST establish a
subscription to the dialog event package of the target endpoint, by
sending a SUBSCRIBE request containing an Event header field set to
"dialog" to the terminating SBE. The originating SBE MUST populate
the SUBSCRIBE Request-URI with the URI returned in the Contact header
field of the INVITE response, if that URI is a GRUU. Otherwise, the
originating SBE MUST populate the Request-URI with the identity of
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the target callback/recall user.
5.6.3. Target Sends NOTIFY to Originating SBE
On receiving the SUBSCRIBE to the dialog event package, the
terminating SBE MUST notify the originating SBE of the dialog state
of the target user endpoint as described in [RFC4235]. Upon
receiving a NOTIFY message of "target is idle", the originating SBE
MUST first cancel the dialog-event subscription by sending a
SUBSCRIBE message with an Expires header containing the value "0".
Once the subscription is cancelled, the originating SBE MUST send a
new INVITE request to establish a call with the target user. If the
originating SBE receives a 486-Busy-Here or 600-Busy-Everywhere
response to the INVITE, then it MUST automatically re-subscribe to
the dialog event package of the target user. (Note, a "busy"
response could be returned in this case as a result of a race
condition, where the target endpoint sends a NOTIFY of "target is
idle", and then becomes busy in a new call before the subsequent
INVITE is received).
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6. Security Considerations
This draft contains no new security considerations that have not
already been defined in SIP and the specified SIP extensions in this
draft.
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7. Acknowledgements
The authors of this draft wish to thank Tom Creighton, Jack Burton,
Matt Cannon, Robert Diande, Jean-Francois Mule, and Kevin Johns for
their contributions to this draft.
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8. IANA Considerations
This draft contains no IANA considerations.
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9. Normative References
[I-D.ietf-sip-gruu]
Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", draft-ietf-sip-gruu-15 (work in progress),
October 2007.
[I-D.ietf-speermint-architecture]
Penno, R. and S. Khan, "SPEERMINT Peering Architecture",
draft-ietf-speermint-architecture-08 (work in progress),
March 2009.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[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 Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3265] Roach, A., "Session Initiation Protocol (SIP)-Specific
Event Notification", RFC 3265, June 2002.
[RFC3323] Peterson, J., "A Privacy Mechanism for the Session
Initiation Protocol (SIP)", RFC 3323, November 2002.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
Extensions to the Session Initiation Protocol (SIP) for
Asserted Identity within Trusted Networks", RFC 3325,
November 2002.
[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[RFC3725] Rosenberg, J., Peterson, J., Schulzrinne, H., and G.
Camarillo, "Best Current Practices for Third Party Call
Control (3pcc) in the Session Initiation Protocol (SIP)",
BCP 85, RFC 3725, April 2004.
[RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
Protocol (SIP) "Replaces" Header", RFC 3891,
September 2004.
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[RFC3966] Schulzrinne, H., "The tel URI for Telephone Numbers",
RFC 3966, December 2004.
[RFC4028] Donovan, S. and J. Rosenberg, "Session Timers in the
Session Initiation Protocol (SIP)", RFC 4028, April 2005.
[RFC4235] Rosenberg, J., Schulzrinne, H., and R. Mahy, "An INVITE-
Initiated Dialog Event Package for the Session Initiation
Protocol (SIP)", RFC 4235, November 2005.
[RFC4244] Barnes, M., "An Extension to the Session Initiation
Protocol (SIP) for Request History Information", RFC 4244,
November 2005.
[RFC4458] Jennings, C., Audet, F., and J. Elwell, "Session
Initiation Protocol (SIP) URIs for Applications such as
Voicemail and Interactive Voice Response (IVR)", RFC 4458,
April 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4694] Yu, J., "Number Portability Parameters for the "tel" URI",
RFC 4694, October 2006.
[RFC5486] Malas, D. and D. Meyer, "Session Peering for Multimedia
Interconnect (SPEERMINT) Terminology", RFC 5486,
March 2009.
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Authors' Addresses
David Hancock
CableLabs
858 Coal Creek Circle
Louisville, CO 80027
USA
Email: d.hancock@cablelabs.com
Daryl Malas
CableLabs
858 Coal Creek Circle
Louisville, CO 80027
USA
Email: d.malas@cablelabs.com
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