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This document proposes the use of the Reason header field in SIP responses.
1.
Terminology
2.
Overview
3.
Requirements
3.1.
Used Cases for the requirements
4.
Overall Applicability
5.
Procedures
5.1.
Procedures at the UA
5.2.
Procedures at a SIP proxy
5.3.
Procedures at an application server
6.
Procedures at an interworking point with ISUP
7.
Example
8.
Security Considerations
9.
IANA Considerations
10.
Acknowledgments
11.
Normative References
§
Authors' Addresses
§
Intellectual Property and Copyright Statements
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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 [RFC2119] (Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” March 1997.).
This document uses terms from [RFC3261] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.).
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The European Telecommunication Standards Institute (ETSI) is defining a Next Generation Network (NGN) where a substantial part of it is based on the IP Multimedia Subsystem (IMS) defined by the Third-Generation Partnership Project (3GPP). IMS is largely based on the Session Initiation Protocol [RFC3261] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.).
ETSI has developed a number of requirements to support the usage of SIP in Next Generation Networks that interoperate, at the service level, with the Public Switched Telephone Network (PSTN), the Integrated Services Digital Network (ISDN), the 3GPP IP Multimedia Subsystem (IMS), and SIP networks and terminals that implement the service logic.
In order to provide full support in SIP of existing services, extensions to SIP are needed.
This document proposes the use of the Reason header field in responses. This is needed for creating services that must be interoperable with the PSTN/ISDN network and the interoperability of traversing communications through SIP not using SIP-I.
The main used case for reason header within responses are interworking situations with PSTN/ISDN networks where the ISUP cause In many cases the mapping of specific cause values will result in a generic SIP Response like it is shown below.
[RFC3398] (Camarillo, G., Roach, A., Peterson, J., and L. Ong, “Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping,” December 2002.) and other Interworking specifications like [3GPP.29.163] (3GPP, “Interworking between the IP Multimedia (IM) Core Network (CN) subsystem and Circuit Switched (CS) networks,” June 2009.) are describing the mapping of ISUP Cause Values to SIP and vice versa. Looking on the specific mapping shows that information will be lost when the call traverses ISUP without using SIP-T.
Example:
[RFC3398] (Camarillo, G., Roach, A., Peterson, J., and L. Ong, “Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping,” December 2002.) describes the mapping of following ISUP Causes to 503 and 408 like follows.
ISUP Cause value SIP response
---------------- ------------
34 no circuit available 503 Service unavailable
38 network out of order 503 Service unavailable
41 temporary failure 503 Service unavailable
42 switching equipment congestion 503 Service unavailable
47 resource unavailable 503 Service unavailable
58 bearer capability not presently 503 Service unavailable
Available
88 incompatible destination 503 Service unavailable
18 no user responding 408 Request Timeout
The mapping back is shown as follows:
Response received Cause value in the REL
----------------- ----------------------
503 Service unavailable 41 Temporary failure
408 Request timeout 102 Recovery on timer
expiry
The Example with 503 shows that a couple of different ISUP Cause values are interworked to only one SIP response. With 408 the meaning of the release cause is changed when interworked back to ISUP. Also Services built on Cause 18 (e.G. a 2nd call attempt on an other number, this service is like a sequential forking) will not work.
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REQ-1:
It should be possible to support PSTN-SIP-PSTN scenarios where the reason of a call release can be transferred though the SIP domain without any loss of information and no change of reason.
REQ-2:
It should be possible to provide correct announcements to a SIP user based on the reason for call clearing within the PSTN network or the PSTN user. The PSTN does normally not provide announcements to originating user when clearing the call.
REQ-3:
A UA may have the ability to display ISUP specific release causes or show a equivalent text. A inclusion of Q.850 causes is out of scope.
REQ-4:
A application server providing specific PSTN like services may have the possibility to include ISUP specific release causes
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REQ-1 shows the scenario between two ISUP Gateways where SIP-I is not a solution to be used. A IMS network defined by 3GPP where SIP-I is not part of the framework is such an example. A used case is shown in Figure 2.
REQ-2 identifies the scenario the gateway passes on the cause and some announcement server translates the cause into the appropriate announcement.
REQ-3 excludes that a SIP UA includes Q.850 cause values because that is not needed due to the different context of a SIP UA and of course the existing SIP response codes. Also the possibility of displaying Q.850 causes should be used very restrictive and is not recommended. A used case could be a Integrated Access Device where a POTS Phone is connected to. It is recommended to ignore the cause value if received be a UA.
REQ-4 allows an application server that is providing specific Services simulating PSTN services to include Q.850 Causes. This is needed specific for IMS operartors which are substituting their PSTN to IMS. These operators may run Services in the SIP domain that are provided to PSTN end users.
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The SIP procedures specified in this document are foreseen for networks providing simulation services and/or interworking to the PSTN/ISDN.
The document is describing the use of the Reason header in SIP responses. These procedures are only valuable if the reason contained in the element "protocol" is "Q.850". A inclusion of a SIP reason (protocol="SIP") is not helpful due to the fact that the response already provides the SIP reason. The Release Causes are described within [ETSI_EN300_485] (ETSI, “Integrated Services Digital Network (ISDN); Definition and usage of cause and location in Digital Subscriber Signalling System No. one (DSS1) and Signalling System No. 7 (SS7) ISDN User Part (ISUP) [ITU-T Recommendation Q.850 (1998) with addendum modified],” .). (Note: The ETSI specifications can be downloaded under http://pda.etsi.org/pda/queryform.asp free of charge.)
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A UA that supports the Reason header field can process the Q.850 Cause Value and display it or an equivalent text. The inclusion of a Reason header field by UA is only for B2B UA interworking with the PSTN/ISDN or providing services foreseen.
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SIP proxies that receive a response containing a Reason header field is forwarding the response without changing the reason.
A SIP proxy receiving a request that includes a Reason header field can route the request to an application server for further analysis and base services on it.
Based on network policy a Proxy can remove a Reason header field send from a UAC.
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An application server that receives a SIP request that contains a response including a Reason header MAY analyze the SIP Reason and base further procedures on this analyses.
For Example the application server could use the reason for sending a announcement towards the originating entity of the session.
As an example the Anonymous Communication Rejection (ACR) service defined by ETSI Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN)
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For interoperability reasons the Q.850 Cause Value of a Release shall be mapped to the Reason Header.
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Figure 1 shows the example of SIP interworking with the PSTN/ISDN. Cause #87 is sent when the connecting user is not member of a Closed User Group.
A Gateway Proxy AS
| IAM | | |
|------------------>| INVITE | |
| |----------------->| INVITE |
| | 100 Trying |----------------->|
| |<-----------------| 100 Trying |
| | |<-----------------|
| ACK SDP held | | |
|<------------------| | 603 Decline |
| | 603 Decline | Reason Q850 #87 |
| | Reason Q850 #87 | |
| REL Cause #87 | |<-----------------|
| |<-----------------| |
|<----------------- | | |
| | | |
| | | |
| | | |
| Figure 1: ISUP-SIP Call |
Figure 2 shows the example where the SIP network is used as transit between PSTN/ISDN networks. This avoids that the Mapping back to the Q.850 cause within ISUP change the meaning of the reason for release of the call.
A Gateway Gateway B
| IAM | | |
|------------------>| INVITE | |
| |----------------->| IAM |
| | 100 Trying |----------------->|
| |<-----------------| 100 Trying |
| | 603 Decline | |
| | Reason Q850 #87 | REL Cause #87 |
| REL Cause #87 | |<-----------------|
| <-----------------| |
|<----------------- | | |
| | | |
| | | |
| | | |
| Figure 2: Transit case |
Figure 3 shows the example where the SIP network puts an announcement towards the UAB. The AS sends an announcement with a specific text back. After some Time the Response will be sent back to the UA A and closes all open transactions. With this possibility the SIP user can informed with more specific information than only the Response code.
A AS Gateway B
| INVITE | | |
|------------------>| INVITE | |
| |----------------->| IAM |
| | 100 Trying |----------------->|
| |<-----------------| |
| | 503 Decline | |
| | Reason Q850 #41 | REL Cause #41 |
| | |<-----------------|
| Announcement |<-----------------| |
|< ================ | | |
| | | |
| 503 after Timeout| | |
|<----------------- | | |
| Figure 3: Transit case |
Figure 3: Call Release within the PSTN with an announce played within the SIP network.
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The presence of the Reason header in a response does not affect the treatment of the response.
Including such a header by an untrusted entity could adulterate the reactions of the originating entities. E.G. sending back a cause value "87" can cause an announcement within the PSTN/ISDN saying that the call was rejected due to the Closed User Group service.
Therefore it is RECOMMENDED to include the Reason header information in Responses only by trusted entities as it is described within [RFC3325] (Jennings, C., Peterson, J., and M. Watson, “Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks,” November 2002.).
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This document describes the use of the Reason header field described within [RFC3326] (Schulzrinne, H., Oran, D., and G. Camarillo, “The Reason Header Field for the Session Initiation Protocol (SIP),” December 2002.) . No additional SIP elements are defined within this document. Therefore, this document does not provide any action to IANA.
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The author would like to thank the members of the ETSI TISPAN WG3 for their comments to this memo.
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| [3GPP.29.163] | 3GPP, “Interworking between the IP Multimedia (IM) Core Network (CN) subsystem and Circuit Switched (CS) networks,” 3GPP TS 29.163 6.12.0, June 2009. |
| [ETSI_EN300_485] | ETSI, “Integrated Services Digital Network (ISDN); Definition and usage of cause and location in Digital Subscriber Signalling System No. one (DSS1) and Signalling System No. 7 (SS7) ISDN User Part (ISUP) [ITU-T Recommendation Q.850 (1998) with addendum modified].” |
| [RFC2119] | Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” BCP 14, RFC 2119, March 1997 (TXT, HTML, XML). |
| [RFC2396] | Berners-Lee, T., Fielding, R., and L. Masinter, “Uniform Resource Identifiers (URI): Generic Syntax,” RFC 2396, August 1998 (TXT, HTML, XML). |
| [RFC2401] | Kent, S. and R. Atkinson, “Security Architecture for the Internet Protocol,” RFC 2401, November 1998 (TXT, HTML, XML). |
| [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 (TXT). |
| [RFC3263] | Rosenberg, J. and H. Schulzrinne, “Session Initiation Protocol (SIP): Locating SIP Servers,” RFC 3263, June 2002 (TXT). |
| [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 (TXT). |
| [RFC3326] | Schulzrinne, H., Oran, D., and G. Camarillo, “The Reason Header Field for the Session Initiation Protocol (SIP),” RFC 3326, December 2002 (TXT). |
| [RFC3398] | Camarillo, G., Roach, A., Peterson, J., and L. Ong, “Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping,” RFC 3398, December 2002 (TXT). |
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| Roland Jesske | |
| Deutsche Telekom | |
| Heinrich-Hertz-Strasse 3-7 | |
| Darmstadt, 64307 | |
| Germany | |
| Phone: | +4961516282766 |
| Email: | r.jesske@telekom.de |
| Martin Hülsemann | |
| Deutsche Telekom | |
| Heinrich-Hertz-Strasse 3-7 | |
| Darmstadt, 64307 | |
| Germany | |
| Phone: | +4961516282765 |
| Email: | martin.huelsemann@telekom.de |
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