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IETF RFC 6405
Last modified on Saturday, November 5th, 2011
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Internet Engineering Task Force (IETF) A. Uzelac, Ed.
Request for Comments: 6405 Global Crossing
Category: Informational Y. Lee, Ed.
ISSN: 2070-1721 Comcast Cable
November 2011
Voice over IP (VoIP) SIP Peering Use Cases
 Abstract
This document depicts many common Voice over IP (VoIP) use cases for
Session Initiation Protocol (SIP) peering. These use cases are
categorized into static and on-demand, and then further sub-
categorized into direct and indirect. These use cases are not an
exhaustive set, but rather the most common use cases deployed today.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/RFC 6405.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Uzelac & Lee Informational PAGE 1 
RFC 6405 VoIP SIP Peering Use Cases November 2011
Table of Contents
1. Introduction ....................................................2
2. Terminology .....................................................3
3. Reference Architecture ..........................................3
4. Contexts of Use Cases ...........................................4
5. Use Cases .......................................................4
5.1. Static Peering Use Cases ...................................5
5.2. Static Direct Peering Use Case .............................5
5.2.1. Administrative Characteristics .....................10
5.2.2. Options and Nuances ................................10
5.3. Static Direct Peering Use Case - Assisting LUF and LRF ....11
5.3.1. Administrative Characteristics .....................12
5.3.2. Options and Nuances ................................12
5.4. Static Indirect Peering Use Case - Assisting LUF and LRF ..12
5.4.1. Administrative Characteristics .....................19
5.4.2. Options and Nuances ................................19
5.5. Static Indirect Peering Use Case ..........................19
5.5.1. Administrative Characteristics .....................20
5.5.2. Options and Nuances ................................21
5.6. On-Demand Peering Use Case ................................21
5.6.1. Administrative Characteristics .....................21
5.6.2. Options and Nuances ................................21
6. Acknowledgments ................................................22
7. Security Considerations ........................................22
8. References .....................................................22
8.1. Normative References ......................................22
8.2. Informative References ....................................23
1. Introduction
This document describes important Voice over IP (VoIP) use cases for
SIP-based [RFC 3261] peering. These use cases are determined by the
Session PEERing for Multimedia INTerconnect (SPEERMINT) working group
and will assist in identifying requirements and other issues to be
considered for future resolution by the working group.
Only use cases related to VoIP are considered in this document.
Other real-time SIP communications use cases, like Instant Messaging
(IM), video chat, and presence are out of scope for this document.
The use cases contained in this document are described as
comprehensive as possible, but should not be considered the exclusive
set of use cases.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
2. Terminology
This document uses terms defined in [RFC 5486]. Please refer to it
for definitions.
3. Reference Architecture
The diagram below provides the reader with a context for the VoIP use
cases in this document. Terms such as SIP Service Provider (SSP),
Lookup Function (LUF), Location Routing Function (LRF), Signaling
Path Border Element (SBE), and Data Path Border Element (DBE) are
defined in [RFC 5486].
The Originating SSP (O-SSP) is the SSP originating a SIP request.
The Terminating SSP (T-SSP) is the SSP terminating the SIP request
originating from the O-SSP. The assisting LUF and LRF Provider offer
LUF and LRF services to the O-SSP. The Indirect SSP (I-SSP) is the
SSP providing indirect peering service(s) to the O-SSP to connect to
the T-SSP.
+--------------------+------------------------+--------------------+
| Originating SSP | Assisting LUF and LRF | Terminating SSP |
| Domain | Provider Domain | Domain |
| | | |
| +-----+ +-----+ | +------+ +------+ | +-----+ +-----+ |
| |O-LUF| |O-LRF| | |A-LUF | | A-LRF| | |T-LUF| |T-LRF| |
| +-----+ +-----+ | +------+ +------+ | +-----+ +-----+ |
| | | |
| +-------+ +-----+ +------------------------+ +-----+ +-------+ |
| |O-Proxy| |O-SBE| | Indirect SSP Domain | |T-SBE| |T-Proxy| |
| +-------+ +-----+ | | +-----+ +-------+ |
| | +-----+ +-----+ | |
| +---+ +-----+ | |O-SBE| |O-DBE| | +-----+ +---+ |
| |UAC| |O-DBE| | +-----+ +-----+ | |T-DBE| |UAS| |
| +---+ +-----+ | | +-----+ +---+ |
| | | |
+--------------------+------------------------+--------------------+
General Overview
Figure 1
Note that some elements included in Figure 1 are optional.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
4. Contexts of Use Cases
Use cases are sorted into two general groups: static and on-demand
peering [RFC 5486]. Each group can be further sub-divided into Direct
Peering and Indirect Peering [RFC 5486]. Although there may be some
overlap among the use cases in these categories, there are different
requirements between the scenarios. Each use case must specify a
basic set of required operations to be performed by each SSP when
peering.
These can include:
o Peer Discovery - Peer discovery via a Lookup Function (LUF) to
determine the Session Establishment Data (SED) [RFC 5486] of the
request. In VoIP use cases, a request normally contains a phone
number. The O-SSP will input the phone number to the LUF and the
LUF will normally return a SIP address of record (AOR) [RFC 3261]
that contains a domain name.
o Next-Hop Routing Determination - Resolving the SED information is
necessary to route the request to the T-SSP. The LRF is used for
this determination. After obtaining the SED, the O-SSP may use
the standard procedure defined in [RFC 3263] to discover the next-
hop address.
o Call setup - SSPs that are interconnecting to one another may also
define specifics on what peering policies need to be used when
contacting the next hop in order to a) reach the next hop at all
and b) prove that the sender is a legitimate peering partner.
Examples: hard-code transport (TCP/UDP/TLS), non-standard port
number, specific source IP address (e.g., in a private Layer 3
network), which TLS client certificate [RFC 5246] to use, and other
authentication schemes.
o Call reception - This step ensures that the type of relationship
(static or on-demand, indirect or direct) is understood and
acceptable. For example, the receiving SBE needs to determine
whether the INVITE it received really came from a trusted member.
5. Use Cases
Please note there are intra-domain message flows within the use cases
to serve as supporting background information. Only inter-domain
communications are germane to this document.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
5.1. Static Peering Use Cases
Static peering [RFC 5486] describes the use case when two SSPs form a
peering relationship with some form of association established prior
to the exchange of traffic. Pre-association is a prerequisite to
static peering. Static peering is used in cases when two peers want
a consistent and tightly controlled approach to peering. In this
scenario, a number of variables, such as an identification method
(remote proxy IP address) and Quality-of-Service (QoS) parameters,
can be defined upfront and known by each SSP prior to peering.
5.2. Static Direct Peering Use Case
This is the simplest form of a peering use case. Two SSPs negotiate
and agree to establish a SIP peering relationship. The peer
connection is statically configured and the peer SSPs are directly
connected. The peers may exchange interconnection parameters such as
Differentiated Service Code Point (DSCP) [RFC 2474] policies, the
maximum number of requests per second, and proxy location prior to
establishing the interconnection. Typically, the T-SSP only accepts
traffic originating directly from the trusted peer.
+--------------------+ +---------------------+
| O-SSP | | T-SSP |
| +-----+ | | +-----+ |
| |O-LUF| | | |T-LUF| |
| |O-LRF| | | /|T-LRF| |
| /+-----+\ | | / +-----+ |
| (2) (4,5,6) | | / |
| / \ | | /(8,9) |
|+-------+ +-----+ +-----+ +-------+|
||O-Proxy|-(3)-|O-SBE+-----(7)-----+T-SBE|-(10)-|T-Proxy||
|+-------+ +-----+ +-----+ +-------+|
| | | | | |
| (1) | | (11) |
| | | | | |
| +-----+ +-----+ +-----+ +-----+ |
| | UAC +======|O-DBE+=====(12)====+T-DBE|=======+ UAS | |
| +-----+ +-----+ +-----+ +-----+ |
+--------------------+ +---------------------+
example.com example.net
Static Direct Peering Use Case
Figure 2
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The following is a high-level depiction of the use case:
1. The User Agent Client (UAC) initiates a call via SIP INVITE to
O-Proxy. O-Proxy is the home proxy for UAC.
INVITE sip:+19175550100@example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.example.com:5060
;branch=z9hG4bK74bf9
Max-Forwards: 10
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=12345
To: Bob <sip:+19175550100@example.com;user=phone>
Call-ID: abcde
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@client.example.com;user=phone;
transport=tcp>
</allOneLine>
Note that UAC inserted its Fully Qualified Domain Name (FQDN) in the
VIA and CONTACT headers. This example assumes that UAC has its own
FQDN.
2. UAC knows the User Agent Server's (UAS's) TN, but does not know
UAS's domain. It appends its own domain to generate the SIP URI
in the Request-URI and TO header. O-Proxy checks the Request-
URI and discovers that the Request-URI contains the user
parameter "user=phone". This parameter signifies that the
Request-URI is a phone number. So O-Proxy will extract the TN
from the Request-URI and query the LUF for SED information from
a routing database. In this example, the LUF is an ENUM
[RFC 6116] database. The ENUM entry looks similar to this:
$ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
IN NAPTR (
10
100
"u"
"E2U+SIP"
"!^.*$!sip:+19175550100@example.net!"
. )
This SED data can be provisioned by O-SSP or populated by the T-SSP.
3. O-Proxy examines the SED and discovers the domain is external.
Given the O-Proxy's internal routing policy, O-Proxy decides to
use O-SBE to reach T-SBE. O-Proxy routes the INVITE request to
O-SBE and adds a Route header that contains O-SBE.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
INVITE sip:+19175550100@example.net;user=phone SIP/2.0
Via: SIP/2.0/TCP o-proxy.example.com:5060
;branch=z9hG4bKye8ad
Via: SIP/2.0/TCP client.example.com:5060
;branch=z9hG4bK74bf9;received=192.0.1.1
Max-Forwards: 9
Route: <sip:o-sbe1.example.com;lr>
Record-Route: <sip:o-proxy.example.com;lr>
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=12345
To: Bob <sip:+19175550100@example.com;user=phone>
Call-ID: abcde
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@client.example.com;user=phone;
transport=tcp>
</allOneLine>
4. O-SBE receives the requests and pops the top entry of the Route
header that contains "o-sbe1.example.com". O-SBE examines the
Request-URI and does an LRF for "example.net". In this example,
the LRF is a Naming Authority Pointer (NAPTR) DNS query
[RFC 3403] of the domain name. O-SBE receives a NAPTR response
from the LRF. The response looks similar to this:
IN NAPTR (
50
50
"S"
"SIP+D2T"
""
_sip._tcp.t-sbe.example.net. )
IN NAPTR (
90
50
"S"
"SIP+D2U"
""
_sip._udp.t-sbe.example.net. )
5. Given the lower order for TCP in the NAPTR response, O-SBE
decides to use TCP as the transport protocol, so it sends an SRV
DNS query for the SRV record [RFC 2782] for "_sip._tcp.t-
sbe.example.net." to O-LRF.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
;; priority weight port target
IN SRV 0 2 5060 t-sbe1.example.net.
IN SRV 0 1 5060 t-sbe2.example.net.
6. Given the higher weight for "t-sbe1.example.net", O-SBE sends an
A record DNS query for "t-sbe1.example.net." to get the A
record:
;; DNS ANSWER
t-sbe1.example.net. IN A 192.0.2.100
t-sbe1.example.net. IN A 192.0.2.101
7. O-SBE sends the INVITE to T-SBE. O-SBE is the egress point to
the O-SSP domain, so it should ensure subsequent mid-dialog
requests traverse via itself. If O-SBE chooses to act as a
back-to-back user agent (B2BUA) [RFC 3261], it will generate a
new INVITE request in next step. If O-SBE chooses to act as a
proxy, it should record-route to stay in the call path. In this
example, O-SBE is a B2BUA.
INVITE sip:+19175550100@example.net;user=phone SIP/2.0
Via: SIP/2.0/TCP o-sbe1.example.com:5060
;branch= z9hG4bK2d4zzz
Max-Forwards: 8
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=54321
To: Bob <sip:+19175550100@example.net;user=phone>
Call-ID: abcde-osbe1
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@o-sbe1.example.com;user=phone;
transport=tcp>
</allOneLine>
Note that O-SBE may re-write the Request-URI with the target domain
in the SIP URI. Some proxy implementations will only accept the
request if the Request-URI contains their own domains.
8. T-SBE determines the called party home proxy and directs the
call to the called party. T-SBE may use ENUM lookup or other
internal mechanism to locate the home proxy. If T-SSP uses ENUM
lookup, this internal ENUM entry is different from the external
ENUM entry populated for O-SSP. In this example, the internal
ENUM query returns the UAS's home proxy.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
$ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
IN NAPTR (
10
100
"u"
"E2U+SIP"
"!^.*$!sip:+19175550100@t-proxy.example.net!"
. )
9. T-SBE receives the NAPTR record, and following the requirements
in [RFC 3263], queries DNS for the SRV records indicated by the
NAPTR result. Not finding any, the T-SBE then queries DNS for
the A record of domain "t-proxy.example.net.".
;; DNS ANSWER
t-proxy.example.net. IN A 192.0.2.2
10. T-SBE is a B2BUA, so it generates a new INVITE and sends it to
UAS's home proxy:
INVITE sip:bob@t-proxy.example.net;user=phone SIP/2.0
Via: SIP/2.0/TCP t-sbe1.example.net:5060
;branch= z9hG4bK28uyyy
Max-Forwards: 7
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=54321
To: Bob <sip:+19175550100@t-proxy.example.net;user=phone>
Call-ID: abcde-tsbe1
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@t-sbe1.example.net;user=phone;
transport=tcp>
</allOneLine>
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RFC 6405 VoIP SIP Peering Use Cases November 2011
11. Finally, UAS's home proxy forwards the INVITE request to the
UAS.
INVITE sip:+19175550100@server.example.net;user=phone SIP/2.0
Via: SIP/2.0/TCP t-proxy.example.net:5060
;branch= z9hG4bK28u111
Via: SIP/2.0/TCP t-sbe1.example.net:5060
;branch= z9hG4bK28uyyy; received=192.2.0.100
Max-Forwards: 6
Record-Route: <sip:t-proxy.example.net:5060;lr>,
<sip:t-sbe1.example.net:5060;lr>
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=54321
To: Bob <sip:+19175550100@t-proxy.example.net;user=phone>
Call-ID: abcde-tsbe1
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@t-sbe1.example.net;user=phone;
transport=tcp>
</allOneLine>
12. RTP is established between the UAC and UAS. Note that the media
shown in Figure 2 passes through O-DBE and T-DBE, but the use of
DBE is optional.
5.2.1. Administrative Characteristics
The static direct peering use case is typically implemented in a
scenario where there is a strong degree of trust between the two
administrative domains. Both administrative domains typically sign a
peering agreement that states clearly the policies and terms.
5.2.2. Options and Nuances
In Figure 2, O-SSP and T-SSP peer via SBEs. Normally, the operator
will deploy the SBE at the edge of its administrative domain. The
signaling traffic will pass between two networks through the SBEs.
The operator has many reasons to deploy an SBE. For example, the
O-SSP may use [RFC 1918] addresses for their UA and proxies. These
addresses are not routable in the target network. The SBE can
perform a NAT function. Also, the SBE eases the operation cost for
deploying or removing Layer 5 network elements. Consider the
deployment architecture where multiple proxies connect to a single
SBE. An operator can add or remove a proxy without coordinating with
the peer operator. The peer operator "sees" only the SBE. As long
as the SBE is maintained in the path, the peer operator does not need
to be notified.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
When an operator deploys SBEs, the operator is required to advertise
the SBE to the peer LRF so that the peer operator can locate the SBE
and route the traffic to the SBE accordingly.
SBE deployment is a decision within an administrative domain. Either
one or both administrative domains can decide to deploy SBE(s). To
the peer network, most important is to identify the next-hop address.
This decision does not affect the network's ability to identify the
next-hop address.
5.3. Static Direct Peering Use Case - Assisting LUF and LRF
This use case shares many properties with the Static Direct Peering
Use Case Section 5.2. There must exist a pre-association between the
O-SSP and T-SSP. The difference is O-SSP will use the Assisting LUF/
LRF Provider for LUF and LRF. The LUF/LRF Provider stores the SED to
reach T-SSP and provides it to O-SSP when O-SSP requests it.
+-----------------+
|LUF/LRF Provider |
| |
| +-------+ |
| +-+ A-LUF | |
| / | A-LRF | |
+--------------------+ / ++-------+ +---------------------+
| O-SSP |/ / | T-SSP |
| +------------/(4,5,6) | +-----+ |
| / | / | |T-LUF| |
| (2) +-+/ | +-|T-LRF| |
| / / | | / +-----+ |
| / / | | /(8,9) |
|+-------+ +-----+ +-----+ +-------+|
||O-Proxy|-(3)-|O-SBE+-------(7)-------+T-SBE|-(10)-|T-Proxy||
|+-------+ +-----+ +-----+ +-------+|
| | | | | |
| (1) | | (11) |
| | | | | |
| +-----+ +-----+ +-----+ +-----+ |
| | UAC +======|O-DBE+=======(12)======+T-DBE+=======+ UAS | |
| +-----+ +-----+ +-----+ +-----+ |
+--------------------+ +---------------------+
example.com example.net
Static Direct Peering with Assisting LUF and LRF
Figure 3
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RFC 6405 VoIP SIP Peering Use Cases November 2011
The call flow looks almost identical to Static Direct Peering Use
Case except that Steps 2, 4, 5, and 6 involve the LUF/LRF Provider
instead of happening in O-SSP domain.
Similar to Static Direct Peering Use Case, the O-DBE and T-DBE in
Figure 3 are optional.
5.3.1. Administrative Characteristics
The LUF/LRF Provider supplies the LUF and LRF services for the O-SSP.
Taken together, the LUF/LRF Provider, O-SSP, and T-SSP form a trusted
administrative domain. To reach the T-SSP, the O-SSP must still
require pre-arranged agreements for the peer relationship with the
T-SSP. The Layer 5 policy is maintained in the O-SSP and T-SSP
domains, and the LUF/LRF Provider may not be aware of any Layer 5
policy between the O-SSP and T-SSP.
A LUF/LRF Provider can serve multiple administrative domains. The
LUF/LRF Provider typically does not share SED from one administrative
domain to another administrative domain without appropriate
permission.
5.3.2. Options and Nuances
The LUF/LRF Provider can use multiple methods to provide SED to the
O-SSP. The most commonly used are an ENUM lookup and a SIP Redirect.
The O-SSP should negotiate with the LUF/LRF Provider regarding which
query method it will use prior to sending a request to the LUF/LRF
Provider.
The LUF/LRF Providers must be populated with the T-SSP's AORs and
SED. Currently, this procedure is non-standardized and labor
intensive. A more detailed description of this problem has been
documented in the work in progress [DRINKS].
5.4. Static Indirect Peering Use Case - Assisting LUF and LRF
The difference between a Static Direct Use Case and a Static Indirect
Use Case lies within the Layer 5 relationship maintained by the O-SSP
and T-SSP. In the Indirect use case, the O-SSP and T-SSP do not have
direct Layer 5 connectivity. They require one or multiple Indirect
Domains to assist with routing the SIP messages and possibly the
associated media.
In this use case, the O-SSP and T-SSP want to form a peer
relationship. For some reason, the O-SSP and T-SSP do not have
direct Layer 5 connectivity. The reasons may vary, for example,
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RFC 6405 VoIP SIP Peering Use Cases November 2011
business demands and/or domain policy controls. Due to this indirect
relationship, the signaling will traverse from the O-SSP through one
or multiple I-SSPs to reach the T-SSP.
In addition, the O-SSP is using a LUF/LRF Provider. This LUF/LRF
Provider stores the T-SSP's SED pre-populated by the T-SSP. One
important motivation to use the LUF/LRF Provider is that the T-SSP
only needs to populate its SED once to the provider. Using an LUF/
LRF Provider allows the T-SSP to populate its SED once, while any
O-SSP T-SSP's SED can use this LUF/LRF Provider. Current practice
has shown that it is rather difficult for the T-SSP to populate its
SED to every O-SSP who must reach the T-SSP's subscribers. This is
especially true in the Enterprise environment.
Note that the LUF/LRF Provider and the I-SSP can be the same provider
or different providers.
+------------------+
| LUF/LRF Provider |
| I-SSP |
| +-------+ |
| ---+ A-LUF | |
| / | A-LRF | |
+--------------------+ / +-------+ +---------------------+
| O-SSP |/ / | T-SSP |
| +-------------/ / | +-----+ |
| / |(4,5,6) | |T-LUF| |
| / | / | +----+T-LRF| |
| (2) + +--- | / +-----+ |
| / / | | /(9,10) |
|+-------+ +-----+ +-----+ +-----+ +-------+|
||O-Proxy|-(3)-|O-SBE+-(7)-+I-SBE+-(8)--+T-SBE+-(11)-|T-Proxy||
|+-------+ +-----+ +-----+ +-----+ +-------+|
| | | | | |
| (1) | | (12) |
| | | | | |
| +-----+ +-----+ +-----+ +-----+ +-----+ |
| | UAC +=(13)=|O-DBE+=====+I-DBE+======+T-DBE+=======+ UAS | |
| +-----+ +-----+ +-----+ +-----+ +-----+ |
+-------------------------------------------------------------+
example.com example.org example.net
Indirect Peering via an LUF/LRF Provider and I-SSP (SIP and Media)
Figure 4
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The following is a high-level depiction of the use case:
1. The UAC initiates a call via SIP INVITE to the O-Proxy. The
O-Proxy is the home proxy for the UAC.
INVITE sip:+19175550100@example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.example.com:5060
;branch=z9hG4bK74bf9
Max-Forwards: 10
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=12345
To: Bob <sip:+19175550100@example.com;user=phone>
Call-ID: abcde
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@client.example.com;user=phone;
transport=tcp>
</allOneLine>
2. The UAC knows the UAS's TN, but does not know the UAS's domain.
It appends its own domain to generate the SIP URI in the
Request-URI and TO header. The O-Proxy checks the Request-URI
and discovers that the Request-URI contains the user parameter
"user=phone". This parameter indicates that the Request-URI is
a phone number. So, the O-Proxy will extract the TN from the
Request-URI and query the LUF for SED information from a routing
database. In this example, the LUF is an ENUM database. The
ENUM entry looks similar to this:
$ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
IN NAPTR (
10
100
"u"
"E2U+SIP"
"!^.*$!sip:+19175550100@example.org!"
. )
Note that the response shows the next-hop is the SBE in I-SSP.
Alternatively, the O-SSP may have a pre-association with the I-SSP.
As such, the O-SSP will forward all requests that contain an external
domain in the Request-URI or an unknown TN to the I-SSP. The O-SSP
will rely on the I-SSP to determine the T-SSP and route the request
correctly. In this configuration, the O-SSP can skip Steps 2, 4, 5,
and 6 and forward the request directly to the I-SBE. This
configuration is commonly used in the Enterprise environment.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
3. Given the O-Proxy's internal routing policy, the O-Proxy decides
to use the O-SBE to reach the I-SBE. The O-Proxy routes the
INVITE request to the O-SBE and adds a Route header that
contains the O-SBE.
INVITE sip:+19175550100@example.org;user=phone SIP/2.0
Via: SIP/2.0/TCP o-proxy.example.com:5060
;branch=z9hG4bKye8ad
Via: SIP/2.0/TCP client.example.com:5060
;branch=z9hG4bK74bf9;received=192.0.1.1
Max-Forwards: 9
Route: <sip:o-sbe1.example.com;lr>
Record-Route: <sip:o-proxy.example.com;lr>
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=12345
To: Bob <sip:+19175550100@example.net;user=phone>
Call-ID: abcde
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@client.example.com;user=phone;
transport=tcp>
</allOneLine>
4. The O-SBE receives the requests and pops the top entry of the
Route header that contains "sip:o-sbe1.example.com". The O-SBE
examines the Request-URI and does an LRF for "example.org". In
this example, the LRF is a NAPTR DNS query of the domain. The
O-SBE receives a response similar to this:
IN NAPTR (
50
50
"S"
"SIP+D2T"
""
_sip._tcp.i-sbe.example.org. )
IN NAPTR (
90
50
"S"
"SIP+D2U"
""
_sip._udp.i-sbe.example.org. )
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RFC 6405 VoIP SIP Peering Use Cases November 2011
5. Given the lower order for TCP in the NAPTR response, the O-SBE
decides to use TCP for transport protocol, so it sends an SRV
DNS query for the SRV record for "_sip._tcp.i-sbe.example.org."
to the O-LRF.
;; priority weight port target
IN SRV 0 2 5060 i-sbe1.example.org.
IN SRV 0 1 5060 i-sbe2.example.org.
6. Given the higher weight for "i-sbe1.example.org", the O-SBE
sends a DNS query for an A record of "i-sbe1.example.org." to
get the A record:
;; DNS ANSWER
i-sbe1.example.org. IN A 192.0.2.200
i-sbe1.example.org. IN A 192.0.2.201
7. The O-SBE sends the INVITE to the I-SBE. The O-SBE is the entry
point to the O-SSP domain, so it should ensure subsequent mid-
dialog requests traverse via itself. If the O-SBE chooses to
act as a B2BUA, it will generate a new back-to-back INVITE
request in the next step. If the O-SBE chooses to act as proxy,
it should record-route to stay in the call path. In this
example, the O-SBE is a B2BUA.
INVITE sip:+19175550100@example.org;user=phone SIP/2.0
Via: SIP/2.0/TCP o-sbe1.example.com:5060
;branch= z9hG4bK2d4zzz
Max-Forwards: 8
Route: <sip:i-sbe1.example.org;lr>
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=54321
To: Bob <sip:+19175550100@example.net;user=phone>
Call-ID: abcde-osbe1
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@o-sbe1.example.com;user=phone;
transport=tcp>
</allOneLine>
8. The I-SBE receives the request and queries its internal routing
database on the TN. It determines that the target belongs to
the T-SSP. Since the I-SBE is a B2BUA, the I-SBE generates a
new INVITE request to the T-SSP.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
INVITE sip:+19175550100@.example.net;user=phone SIP/2.0
Via: SIP/2.0/TCP i-sbe1.example.org:5060
;branch= z9hG4bK2d4777
Max-Forwards: 7
Route: <sip:t-sbe1.example.net;lr>
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=54321
To: Bob <sip:+19175550100@example.net;user=phone>
Call-ID: abcde-isbe1
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@i-sbe1.example.org;user=phone;
transport=tcp>
</allOneLine>
Note that if the I-SSP wants the media to traverse through the I-DBE,
the I-SBE must modify the Session Description Protocol (SDP) in the
Offer to point to its DBE.
9. The T-SBE determines the called party home proxy and directs the
call to the called party. The T-SBE may use ENUM lookup or
another internal mechanism to locate the home proxy. If the
T-SSP uses ENUM lookup, this internal ENUM entry is different
from the external ENUM entry populated for O-SSP. This internal
ENUM entry will contain the information to identify the next hop
to reach the called party. In this example, the internal ENUM
query returns the UAS's home proxy.
$ORIGIN 0.0.1.0.5.5.5.7.1.9.1.e164.arpa.
IN NAPTR (
10
100
"u"
"E2U+SIP"
"!^.*$!sip:+19175550100@t-proxy.example.net!"
. )
Note that this step is optional. If the T-SBE has other ways to
locate the UAS home proxy, the T-SBE can skip this step and send the
request to the UAS's home proxy. We show this step to illustrate one
of the many possible ways to locate UAS's home proxy.
10. The T-SBE receives the NAPTR record and, following the
requirements in [RFC 3263], queries the DNS for the SRV records
indicated by the NAPTR result. Not finding any, the T-SBE then
queries DNS for the A record of domain "t-proxy.example.net.".
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RFC 6405 VoIP SIP Peering Use Cases November 2011
;; DNS ANSWER
t-proxy.example.net. IN A 192.0.2.2
11. T-SBE sends the INVITE to UAS's home proxy:
INVITE sip:+19175550100@t-proxy.example.net;user=phone SIP/2.0
Via: SIP/2.0/TCP t-sbe1.example.net:5060
;branch= z9hG4bK28uyyy
Max-Forwards: 6
Record-Route: <sip:t-sbe1.example.net:5060;lr>
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=54321
To: Bob <sip:+19175550100@example.net;user=phone>
Call-ID: abcde-tsbe1
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@t-sbe1.example.com;user=phone;
transport=tcp>
</allOneLine>
12. Finally, the UAS's home proxy forwards the INVITE request to the
UAS.
INVITE sip:+19175550100@server.example.net;user=phone SIP/2.0
Via: SIP/2.0/TCP t-proxy.example.net:5060
;branch= z9hG4bK28u111
Via: SIP/2.0/TCP t-sbe1.example.net:5060
;branch= z9hG4bK28uyyy; received=192.2.0.100
Max-Forwards: 5
Record-Route: <sip:t-proxy.example.net:5060;lr>,
<sip:t-sbe1.example.net:5060;lr>
From: Alice <sip:+14085550101@example.com;user=phone>
;tag=54321
To: Bob <sip:+19175550100@example.net;user=phone>
Call-ID: abcde-tsbe1
CSeq: 1 INVITE
<allOneLine>
Contact: <sip:+19175550100@t-sbe1.example.com;user=phone;
transport=tcp>
</allOneLine>
13. In Figure 4, RTP is established between the UAC and UAS via the
O-DBE, I-DBE and T-DBE. The use of DBE is optional.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
5.4.1. Administrative Characteristics
This use case looks very similar to the Static Direct Peering Use
Case with Assisting LUF and LRF. The major difference is the O-SSP
and T-SSP do not have direct Layer 5 connectivity. Instead, O-SSP
connects to the T-SSP indirectly via the I-SSP.
Typically, an LUF/LRF Provider serves multiple O-SSPs. Two O-SSPs
may use different I-SSPs to reach the same T-SSP. For example,
O-SSP1 may use I-SSP1 to reach T-SSP, but O-SSP2 may use I-SSP2 to
reach T-SSP. Given the O-SSP and T-SSP pair as input, the LUF/LRF
Provider will return the SED of I-SSP that is trusted by O-SSP to
forward the request to T-SSP.
In this use case, there are two levels of trust relationship. The
first trust relationship is between the O-SSP and the LUF/LRF
Provider. The O-SSP trusts the LUF/LRF to provide the T-SSP's SED.
The second trust relationship is between the O-SSP and I-SSP. The
O-SSP trusts the I-SSP to provide Layer 5 connectivity to assist the
O-SSP in reaching the T-SSP. The O-SSP and I-SSP have a pre-arranged
agreement for policy. Note that Figure 4 shows a single provider to
supply both LUF/LRF and I-SSP, but O-SSP can choose two different
providers.
5.4.2. Options and Nuances
Similar to the Static Direct Peering Use Case, the O-SSP and T-SSP
may deploy SBE and DBE for NAT traversal, security, transcoding, etc.
The I-SSP can also deploy the SBE and DBE for similar reasons (as
depicted in Figure 4).
5.5. Static Indirect Peering Use Case
This use case shares many properties with the Static Indirect Use
Case with Assisting LUF and LRF. The difference is that the O-SSP
uses its internal LUF/LRF to control the routing database. By
controlling the database, the O-SSP can apply different routing rules
and policies to different T-SSPs. For example, the O-SSP can use
I-SSP1 and Policy-1 to reach T-SSP1, and use I-SSP2 and Policy-2 to
reach T-SSP2. Note that there could be multiple I-SSPs and multiple
SIP routes to reach the same T-SSP; the selection process is out of
scope of this document.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
+--------------------+-------------------+---------------------+
| O-SSP | I-SSP | T-SSP |
| +-----+ | | +-----+ |
| -+O-LUF| | | |T-LUF| |
| / |O-LRF+\ | | +----+T-LRF| |
| / +-----+ \ | | / +-----+ |
| /(2) \(4,5,6) | /(9,10) |
|+-------+ +-----+ +-----+ +-----+ +-------+|
||O-Proxy|-(3)-|O-SBE+--(7)-+I-SBE+-(8)--+T-SBE+-(11)-|T-Proxy||
|+-------+ +-----+ +-----+ +-----+ +-------+|
| | | | | |
| (1) | | (12) |
| | | | | |
| +-----+ +-----+ +-----+ +-----+ +-----+ |
| | UAC +=(13)=+O-DBE+======+I-DBE+======+T-DBE+=======+ UAS | |
| +-----+ +-----+ +-----+ +-----+ +-----+ |
+--------------------------------------------------------------+
example.com example.org example.net
Indirect Peering via I-SSP (SIP and Media)
Figure 5
5.5.1. Administrative Characteristics
The Static Indirect Peering Use Case is implemented in cases where no
direct interconnection exists between the originating and terminating
domains due to either business or physical constraints.
O-SSP <---> I-SSP = Relationship O-I
In the O-I relationship, typical policies, features, or functions
that deem this relationship necessary are number portability,
ubiquity of termination options, security certificate management, and
masquerading of originating VoIP network gear.
T-SSP <---> I-SSP = Relationship T-I
In the T-I relationship, typical policies, features, or functions
observed consist of codec "scrubbing", anonymizing, and transcoding.
The I-SSP must record-route and stay in the signaling path. The
T-SSP will not accept any message sent directly from the O-SSP.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
5.5.2. Options and Nuances
In Figure 5, we show an I-DBE. Using an I-DBE is optional. For
example, the I-DBE can be used when the O-SSP and T-SSP do not have a
common codec. To involve an I-DBE, the I-SSP should know the list of
codecs supported by the O-SSP and T-SSP. When the I-SBE receives the
INVITE request, it will make a decision to invoke the I-DBE. An
I-DBE may also be used if the O-SSP uses Secure Real-time Transport
Protocol (SRTP) [RFC 3711] for media and T-SSP does not support SRTP.
5.6. On-Demand Peering Use Case
On-demand peering [RFC 5486] describes how two SSPs form the peering
relationship without a pre-arranged agreement.
The basis of this use case is built on the fact that there is no pre-
established relationship between the O-SSP and T-SSP. The O-SSP and
T-SSP do not share any information prior to the dialog initiation
request. When the O-Proxy invokes the LUF and LRF on the Request-
URI, the terminating user information must be publicly available.
When the O-Proxy routes the request to the T-Proxy, the T-Proxy must
accept the request without any pre-arranged agreement with the O-SSP.
The On-demand Peering Use Case is uncommon in production. In this
memo, we capture only the high-level descriptions. Further analysis
is expected when this use case becomes more popular.
5.6.1. Administrative Characteristics
The On-demand Direct Peering Use Case is typically implemented in a
scenario where the T-SSP allows any O-SSP to reach its serving
subscribers. The T-SSP administrative domain does not require any
pre-arranged agreement to accept the call. The T-SSP makes its
subscribers information publicly available. This model mimics the
Internet email model. The sender does not need an pre-arranged
agreement to send email to the receiver.
5.6.2. Options and Nuances
Similar to the Static Direct Peering Use Case, the O-SSP and T-SSP
can decide to deploy the SBE. Since the T-SSP is open to the public,
the T-SSP is considered to be a higher security risk than static
model because there is no trusted relationship between the O-SSP and
T-SSP. The T-SSP should protect itself from any attack launched by
an untrusted O-SSP.
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RFC 6405 VoIP SIP Peering Use Cases November 2011
6. Acknowledgments
Michael Haberler, Mike Mammer, Otmar Lendl, Rohan Mahy, David
Schwartz, Eli Katz and Jeremy Barkan are the authors of the early
individual documents. Their use cases are captured in this document.
Also, Jason Livingood, Daryl Malas, David Meyer, Hadriel Kaplan, John
Elwell, Reinaldo Penno, Sohel Khan, James McEachern, Jon Peterson,
Alexander Mayrhofer, and Jean-Francois Mule made many valuable
comments related to this document. The editors would also like to
extend a special thank you to Spencer Dawkins for his detailed review
of this document.
7. Security Considerations
Session interconnect for VoIP, as described in this document, has a
wide variety of security issues that should be considered. For
example, if the O-SSP and T-SSP peer through public Internet, the
O-SSP must protect the signaling channel and accept messages only
from an authorized T-SSP. This document does not analyze the threats
in detail. [RFC 6404] discusses the different security threats and
countermeasures related to VoIP peering.
8. References
8.1. Normative References
[RFC 1918] Rekhter, Y., Moskowitz, R., Karrenberg, D., Groot, G., and
E. Lear, "Address Allocation for Private Internets",
BCP 5, RFC 1918, February 1996.
[RFC 2782] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for
specifying the location of services (DNS SRV)", RFC 2782,
February 2000.
[RFC 3261] 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.
[RFC 3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263,
June 2002.
[RFC 3403] Mealling, M., "Dynamic Delegation Discovery System (DDDS)
Part Three: The Domain Name System (DNS) Database",
RFC 3403, October 2002.
Uzelac & Lee Informational PAGE 22
RFC 6405 VoIP SIP Peering Use Cases November 2011
[RFC 5486] Malas, D. and D. Meyer, "Session Peering for Multimedia
Interconnect (SPEERMINT) Terminology", RFC 5486,
March 2009.
[RFC 6116] Bradner, S., Conroy, L., and K. Fujiwara, "The E.164 to
Uniform Resource Identifiers (URI) Dynamic Delegation
Discovery System (DDDS) Application (ENUM)", RFC 6116,
March 2011.
[RFC 6404] Seedorf, J., Niccolini, S., Chen, E., and H. Scholz,
"Session PEERing for Multimedia INTerconnect (SPEERMINT)
Security Threats and Suggested Countermeasures", RFC 6404,
November 2011.
8.2. Informative References
[DRINKS] Channabasappa, S., "Data for Reachability of Inter/
tra-NetworK SIP (DRINKS) Use cases and Protocol
Requirements", Work in Progress, August 2011.
[RFC 2474] Nichols, K., Blake, S., Baker, F., and D. Black,
"Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers", RFC 2474,
December 1998.
[RFC 3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC 5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
Authors' Addresses
Adam Uzelac (editor)
Global Crossing
U.S.A.
EMail: adam.uzelac@globalcrossing.com
URI: http://www.globalcrossing.com
Yiu L.Lee (editor)
Comcast Cable
U.S.A.
EMail: yiu_lee@cable.comcast.com
URI: http://www.comcast.com
Uzelac & Lee Informational PAGE 23
RFC TOTAL SIZE: 51354 bytes
PUBLICATION DATE: Saturday, November 5th, 2011
LEGAL RIGHTS: The IETF Trust (see BCP 78)
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