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<pre>Network Working Group G. Camarillo
Request for Comments: 3578 Ericsson
Category: Standards Track A. B. Roach
dynamicsoft
J. Peterson
NeuStar
L. Ong
Ciena
August 2003
<span class="h1">Mapping of Integrated Services Digital Network (ISDN)</span>
<span class="h1">User Part (ISUP) Overlap Signalling</span>
<span class="h1">to the Session Initiation Protocol (SIP)</span>
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
This document describes a way to map Integrated Services Digital
Network User Part (ISUP) overlap signalling to Session Initiation
Protocol (SIP). This mechanism might be implemented when using SIP
in an environment where part of the call involves interworking with
the Public Switched Telephone Network (PSTN).
<span class="grey">Camarillo, et al. Standards Track [Page 1]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-2" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
Table of Contents
<a href="#section-1">1</a>. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-3">3</a>
2. Conversion of ISUP Overlap Signalling into SIP en-bloc
Signalling . . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-3">3</a>
<a href="#section-2.1">2.1</a>. Waiting for the Minimum Amount of Digits . . . . . . . . <a href="#page-4">4</a>
<a href="#section-2.2">2.2</a>. The Minimum Amount of Digits has been Received . . . . . <a href="#page-4">4</a>
<a href="#section-3">3</a>. Sending Overlap Signalling to a SIP Network. . . . . . . . . . <a href="#page-5">5</a>
<a href="#section-3.1">3.1</a>. One vs. Several Transactions . . . . . . . . . . . . . . <a href="#page-5">5</a>
<a href="#section-3.2">3.2</a>. Generating Multiple INVITEs. . . . . . . . . . . . . . . <a href="#page-6">6</a>
<a href="#section-3.3">3.3</a>. Receiving Multiple Responses . . . . . . . . . . . . . . <a href="#page-8">8</a>
<a href="#section-3.4">3.4</a>. Canceling Pending INVITE Transactions. . . . . . . . . . <a href="#page-9">9</a>
<a href="#section-3.5">3.5</a>. SIP to ISUP. . . . . . . . . . . . . . . . . . . . . . . <a href="#page-9">9</a>
<a href="#section-4">4</a>. Security Considerations. . . . . . . . . . . . . . . . . . . . <a href="#page-10">10</a>
<a href="#section-5">5</a>. Acknowledgments. . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-10">10</a>
<a href="#section-6">6</a>. Normative References . . . . . . . . . . . . . . . . . . . . . <a href="#page-10">10</a>
<a href="#section-7">7</a>. Intellectual Property Statement. . . . . . . . . . . . . . . . <a href="#page-11">11</a>
<a href="#section-8">8</a>. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . <a href="#page-12">12</a>
<a href="#section-9">9</a>. Full Copyright Statement . . . . . . . . . . . . . . . . . . . <a href="#page-13">13</a>
<span class="grey">Camarillo, et al. Standards Track [Page 2]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-3" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>. Introduction</span>
A mapping between the Session Initiation Protocol (SIP) [<a href="#ref-1" title=""SIP: Session Initiation Protocol"">1</a>] and the
ISDN User Part (ISUP) [<a href="#ref-2" title=""Application of the ISDN user part of CCITT signaling system no. 7 for international ISDN interconnections"">2</a>] of SS7 is described in <a href="./rfc3398">RFC 3398</a> [<a href="#ref-3" title=""Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping"">3</a>].
However, <a href="./rfc3398">RFC 3398</a> only takes into consideration ISUP en-bloc
signalling. En-bloc signalling consists of sending the complete
telephone number of the callee in the first signalling message.
Although modern switches always use en-bloc signalling, some parts of
the PSTN still use overlap signalling.
Overlap signalling consists of sending only some digits of the
callee's number in the first signalling message. Further digits are
sent in subsequent signalling messages. Although overlap signalling
in the PSTN is the source of much additional complexity, it is still
in use in some countries.
Like modern switches, SIP uses en-bloc signalling. The Request-URI
of an INVITE request always contains the whole address of the callee.
Native SIP end-points never generate overlap signalling.
Therefore, the preferred solution for a gateway handling PSTN overlap
signalling and SIP is to convert the PSTN overlap signalling into SIP
en-bloc signalling using number analysis and timers. The gateway
waits until all the signalling messages carrying parts of the
callee's number arrive, and only then, it generates a SIP INVITE
request. <a href="#section-2">Section 2</a> describes how to convert ISUP overlap signalling
into en-bloc SIP this way.
However, although it is the preferred solution, conversion of overlap
to en-bloc signalling sometimes results in unacceptable (multiple
second) call setup delays to human users. In these situations, some
form of overlap signalling has to be used in the SIP network to
minimize the call setup delay. However, introducing overlap
signalling in SIP introduces complexity and brings some issues.
<a href="#section-3">Section 3</a> analyzes the issues related to the use of overlap
signalling in a SIP network and describe ways to deal with them in
some particular network scenarios. <a href="#section-3">Section 3</a> also describes in which
particular network scenarios those issues make the use of overlap
signalling in the SIP network unacceptable.
<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>. Conversion of ISUP Overlap Signalling into SIP en-bloc Signalling</span>
In this scenario, the gateway receives an IAM (Initial Address
Message) that contains only a portion of the called number. The rest
of the digits dialed arrive later in one or more SAMs (Subsequent
Address Message).
<span class="grey">Camarillo, et al. Standards Track [Page 3]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-4" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
<span class="h3"><a class="selflink" id="section-2.1" href="#section-2.1">2.1</a>. Waiting for the Minimum Amount of Digits</span>
If the IAM contains less than the minimum amount of digits to route a
call, the gateway starts T35 and waits until the minimum amount of
digits that can represent a telephone number is received (or a stop
digit is received). If T35 expires before the minimum amount of
digits (or a stop digit) has been received, a REL with cause value 28
is sent to the ISUP side. T35 is defined in Q.764 [<a href="#ref-4" title=""Signalling system no. 7 - ISDN user part signalling procedures,"">4</a>] as 15-20
seconds.
If a stop digit is received, the gateway can already generate an
INVITE request with the complete called number. Therefore, the call
proceeds as usual.
<span class="h3"><a class="selflink" id="section-2.2" href="#section-2.2">2.2</a>. The Minimum Amount of Digits has been Received</span>
Once the minimum amount of digits that can represent a telephone
number has been received, the gateway should use number analysis to
decide if the number that has been received so far is a complete
number. If it is, the gateway can generate an INVITE request with
the complete called number. Therefore, the call proceeds as usual.
However, there are cases when the gateway cannot know whether the
number received is a complete number or not. In this case, the
gateway should collect digits until a timer (T10) expires or a stop
digit (such as, #) is entered by the user (note that T10 is refreshed
every time a new digit is received).
When T10 expires, an INVITE with the digits collected so far is sent
to the SIP side. After this, any SAM received is ignored.
PSTN MGC/MG SIP
| | |
|-----------IAM----------->| Starts T10 |
| | |
|-----------SAM----------->| Starts T10 |
| | |
|-----------SAM----------->| Starts T10 |
| | |
| | |
| T10 expires |---------INVITE---------->|
| | |
Figure 1: Use of T10 to convert overlap signalling to en-bloc
<span class="grey">Camarillo, et al. Standards Track [Page 4]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-5" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
Note that T10 is defined for conversion between overlap signalling
(e.g., CAS) and en-bloc ISUP. PSTN switches usually implement a
locally defined value of timer T10 -- which may not be within the 4-6
second range recommended by Q.764 [<a href="#ref-4" title=""Signalling system no. 7 - ISDN user part signalling procedures,"">4</a>] -- to convert overlap ISUP to
en-bloc ISUP. This document uses T10 and recommends the range of
values defined in Q.764 [<a href="#ref-4" title=""Signalling system no. 7 - ISDN user part signalling procedures,"">4</a>], which seems suitable for conversion from
overlap to en-bloc SIP operation. The actual choice of the timer
value is a matter of local policy.
<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>. Sending Overlap Signalling to a SIP Network</span>
This section analyzes the issues related to the use of overlap
signalling in a SIP network and describes a possible solution and its
applicability scope. It is important to note that, if used outside
its applicability scope, this solution could cause a set of problems,
which are identified in this section.
<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>. One vs. Several Transactions</span>
An ingress gateway receiving ISUP overlap signalling (i.e., one IAM
and one or more SAMs) needs to map it into SIP signalling. One
possible approach would consists of sending an INVITE with the digits
received in the IAM, and once an early dialog is established, sending
the digits received in SAMs in a SIP request (e.g., INFO) within that
early dialog.
This approach has several problems. It requires that the remote SIP
user agent (which might be a gateway) sends a non-100 provisional
response as soon as it receives the initial INVITE to establish the
early dialog. Current gateways, following the procedures in <a href="./rfc3398">RFC 3398</a>
[<a href="#ref-3" title=""Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping"">3</a>], do not generate such a provisional response. Having gateways
generate such a response (e.g., 183 Session Progress) would cause
ingress gateways to generate early ACMs, confusing the PSTN state
machine even in calls that do not use overlap signalling.
In this approach, once the initial INVITE request is routed, all the
subsequent requests sent within the early dialog follow the same
path. That is, they cannot be re-routed to take advantage of SIP-
based services. Therefore, we do not recommend using this approach.
An alternative approach consists of sending a new INVITE that
contains all the digits received so far every time a new SAM is
received. Since every new INVITE sent represents a new transaction,
they can be routed in different ways. This way, every new INVITE can
take advantage of any SIP service that the network may provide.
<span class="grey">Camarillo, et al. Standards Track [Page 5]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-6" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
However, having subsequent INVITEs routed in different ways brings
some problems as well. The first INVITE, for instance, might be
routed to a particular gateway, and a subsequent INVITE, to another.
The result is that both gateways generate an IAM. Since one of the
IAMs (or both) has an incomplete number, it would fail, having
already consumed PSTN resources. It could even happen that both IAMs
contained complete, but different numbers (i.e., one number is the
prefix of the other one).
Routing in SIP can be controlled by the administrator of the network.
Therefore, a gateway can be configured to generate SIP overlap
signalling in the way described below only if the SIP routing
infrastructure ensures that INVITEs will only reach one gateway.
When the routing infrastructure is not under the control of the
administrator of the gateway, the procedures of <a href="#section-2">Section 2</a> have to be
used instead.
Within some dialing plans in the PSTN, a phone number might be a
prefix of another one. This situation is not common, but it can
occur. Where en-bloc signalling is used, this ambiguity is resolved
before the digits are placed in the en-bloc signalling. If overlap
signaling was used in this situation, a different user than the one
the caller intended to call might be contacted. That is why in the
parts of the PSTN where overlap is used, a prefix of a telephone
number never identifies another valid number. Therefore, SIP overlap
signalling should not be used when attempting to reach parts of the
PSTN where it is possible for a number and some shorter prefix of the
same number to both be valid addresses of different terminals.
<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>. Generating Multiple INVITEs</span>
In this scenario, the gateway receives an IAM (Initial Address
Message) and possibly one or more SAMs (Subsequent Address Message)
that provide more than the minimum amount of digits that can
represent a phone number.
As soon as the minimum amount of digits is received, the gateway
sends an INVITE and starts T10. This INVITE is built following the
procedures described in <a href="./rfc3398">RFC 3398</a> [<a href="#ref-3" title=""Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping"">3</a>].
If a SAM arrives to the gateway, T10 is refreshed and a new INVITE
with the new digits received is sent. The new INVITE has the same
Call-ID and the same From header field including the tag as the first
INVITE sent, but has an updated Request-URI. The new Request-URI
contains all the digits received so far. The To header field of the
new INVITE contains all the digits as well, but has no tag.
<span class="grey">Camarillo, et al. Standards Track [Page 6]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-7" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
Note that it is possible to receive a response to the first INVITE
before having sent the second INVITE. In this case, the response
received would contain a To tag and information (Record-Route and
Contact) to build a Route header field. The new INVITE to be sent
(containing new digits) should not use any of these headers. That
is, the new INVITE does not contain neither To tag nor Route
header field. This way, this new INVITE can be routed dynamically
by the network providing services.
The new INVITE should, of course, contain a Cseq field. It is
recommended that the Cseq of the new INVITE is higher than any of the
previous Cseq that the gateway has generated for this Call-ID (no
matter for which dialog the Cseq was generated).
When an INVITE forks, responses from different locations might
arrive establishing one or more early dialogs. New requests such
as, PRACK or UPDATE can be sent within every particular early
dialog. This implies that the Cseq number spaces of different
early dialogs are different. Sending a new INVITE with a Cseq
that is still unused by any of the remote destinations avoids
confusion at the destination.
If the gateway is encapsulating ISUP messages as SIP bodies, it
should place the IAM and all the SAMs received so far in this INVITE.
PSTN MGC/MG SIP
| | |
|-----------IAM----------->| Starts T10 |
| |---------INVITE---------->|
| | |
|-----------SAM----------->| Starts T10 |
| |---------INVITE---------->|
| | |
|-----------SAM----------->| Starts T10 |
| |---------INVITE---------->|
| | |
Figure 2: Overlap signalling in SIP
<span class="grey">Camarillo, et al. Standards Track [Page 7]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-8" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
If 4xx, 5xx or 6xx final responses arrive (e.g., 484 address
incomplete) for the pending INVITE transactions before T10 has
expired, the gateway should not send any REL. A REL is sent only if
no more SAMs arrive, T10 expires, and all the INVITEs sent have been
answered with a final response (different than 200 OK).
PSTN MGC/MG SIP
| | |
|-----------IAM----------->| Starts T10 |
| |---------INVITE---------->|
| |<---------484-------------|
| |----------ACK------------>|
| | |
| | |
| T10 expires | |
|<----------REL------------| |
Figure 3: REL generation when overlap signalling is used
The best status code among all the responses received for all the
INVITEs that were generated is used to calculate the cause value of
the REL as described in <a href="./rfc3398">RFC 3398</a> [<a href="#ref-3" title=""Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping"">3</a>].
The computation of the best response is done in the same way as
forking proxies compute the best response to be returned to the
client for a particular INVITE. Note that the best response is
not always the response to the INVITE that contained more digits.
If the user dials a particular number and then types an extra
digit by mistake, a 486 (Busy Here) could be received for the
first INVITE and a 484 (Address Incomplete) for the second one
(which contained more digits).
<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>. Receiving Multiple Responses</span>
When overlap signalling in SIP is used, the ingress gateway sends
multiple INVITEs. Accordingly, it will receive multiple responses.
The responses to all the INVITEs sent, except for one (normally, but
not necessarily the last one), are typically 400 class responses
(e.g., 484 Address Incomplete) that terminate the INVITE transaction.
However, a 183 Session Progress response with a media description can
also be received. The media stream will typically contain a message
such as, "The number you have just dialed does not exist".
<span class="grey">Camarillo, et al. Standards Track [Page 8]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-9" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
The issue of receiving different 183 Session Progress responses with
media descriptions does not only apply to overlap signalling. When
vanilla SIP is used, several responses can also arrive to a gateway
if the INVITE forked. It is then up to the gateway to decide which
media stream should be played to the user.
However, overlap signalling adds a requirement to this process. As a
general rule, a media stream corresponding to the response to an
INVITE with a greater number of digits should be given more priority
than media streams from responses with less digits.
<span class="h3"><a class="selflink" id="section-3.4" href="#section-3.4">3.4</a>. Canceling Pending INVITE Transactions</span>
When a gateway sends a new INVITE containing new digits, it should
not CANCEL the previous INVITE transaction. This CANCEL could arrive
before the new INVITE to an egress gateway and trigger a REL before
the new INVITE arrived. INVITE transactions are typically terminated
by the reception of 4xx responses.
However, once a 200 OK response has been received, the gateway should
CANCEL all the other INVITE transactions were generated. A
particular gateway might implement a timer to wait for some time
before sending any CANCEL. This gives time to all the previous
INVITE transactions to terminate smoothly without generating more
signalling traffic (CANCEL messages).
<span class="h3"><a class="selflink" id="section-3.5" href="#section-3.5">3.5</a>. SIP to ISUP</span>
In this scenario (the call originates in the SIP network), the
gateway receives multiple INVITEs that have the same Call-ID but have
different Request-URIs. Upon reception of the first INVITE, the
gateway generates an IAM following the procedures described in <a href="./rfc3398">RFC</a>
<a href="./rfc3398">3398</a> [<a href="#ref-3" title=""Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping"">3</a>].
When a gateway receives a subsequent INVITE with the same Call-ID and
From tag as the previous one, and an updated Request-URI, a SAM
should be generated as opposed to a new IAM. Upon reception of a
subsequent INVITE, the INVITE received previously is answered with
484 Address Incomplete.
If the gateway is attached to the PSTN in an area where en-bloc
signalling is used, a REL for the previous IAM and a new IAM should
be generated.
<span class="grey">Camarillo, et al. Standards Track [Page 9]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-10" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>. Security Considerations</span>
When overlap signaling is employed, it is possible that an attacker
could send multiple INVITEs containing an incomplete address to the
same gateway in an attempt to occupy all available ports and thereby
deny service to legitimate callers. Since none of these partially
addressed calls would ever complete, in a traditional billing scheme,
the sender of the INVITEs might never be charged. To address this
threat, the authors recommend that gateway operators authenticate the
senders of INVITE requests, first, in order to have some
accountability for the source of calls (it is very imprudent to give
gateway access to unknown users on the Internet), but second, so that
the gateway can determine when multiple calls are originating from
the same source in a short period of time. Some sort of threshold of
hanging overlap calls should be tracked by the gateway, and after the
limit is exceeded, the further similar calls should be rejected to
prevent the saturation of gateway trunking resources.
<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>. Acknowledgments</span>
Jonathan Rosenberg, Olli Hynonen, and Mike Pierce provided useful
feedback on this document.
<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>. Normative References</span>
[<a id="ref-1">1</a>] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
Session Initiation Protocol", <a href="./rfc3261">RFC 3261</a>, June 2002.
[<a id="ref-2">2</a>] "Application of the ISDN user part of CCITT signaling system no.
7 for international ISDN interconnections", ITU-T Q.767,
February 1991.
[<a id="ref-3">3</a>] Camarillo, G., Roach, A. B., Peterson, J. and L. Ong,
"Integrated Services Digital Network (ISDN) User Part (ISUP) to
Session Initiation Protocol (SIP) Mapping", <a href="./rfc3398">RFC 3398</a>, December
2002.
[<a id="ref-4">4</a>] "Signalling system no. 7 - ISDN user part signalling
procedures," ITU-T Q.764, December 1999.
<span class="grey">Camarillo, et al. Standards Track [Page 10]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-11" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>. Intellectual Property Statement</span>
The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it
has made any effort to identify any such rights. Information on the
IETF's procedures with respect to rights in standards-track and
standards-related documentation can be found in <a href="https://www.rfc-editor.org/bcp/bcp11">BCP-11</a>. Copies of
claims of rights made available for publication and any assurances of
licenses to be made available, or the result of an attempt made to
obtain a general license or permission for the use of such
proprietary rights by implementors or users of this specification can
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The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice
this standard. Please address the information to the IETF Executive
Director.
<span class="grey">Camarillo, et al. Standards Track [Page 11]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-12" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>. Authors' Addresses</span>
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
EMail: Gonzalo.Camarillo@ericsson.com
Adam Roach
dynamicsoft
5100 Tennyson Parkway
Suite 1200
Plano, TX 75024
USA
EMail: adam@dynamicsoft.com
Jon Peterson
NeuStar, Inc.
1800 Sutter St
Suite 570
Concord, CA 94520
USA
EMail: jon.peterson@neustar.biz
Lyndon Ong
Ciena
5965 Silver Creek Valley Road
San Jose, CA 95138
USA
EMail: lyong@ciena.com
<span class="grey">Camarillo, et al. Standards Track [Page 12]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-13" ></span>
<span class="grey"><a href="./rfc3578">RFC 3578</a> ISUP Overlap Signalling to SIP August 2003</span>
<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>. Full Copyright Statement</span>
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
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included on all such copies and derivative works. However, this
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The limited permissions granted above are perpetual and will not be
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This document and the information contained herein is provided on an
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TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
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HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
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Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
Camarillo, et al. Standards Track [Page 13]
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