RFC 4497
Interworking between the Session Initiation Protocol (SIP) and QSIG
8. Message Mapping Requirements
8.1. Message Validation and Handling of Protocol Errors
The gateway SHALL validate received QSIG messages in accordance with the requirements of [2] and SHALL act in accordance with [2] on detection of a QSIG protocol error. The requirements of this section for acting on a received QSIG message apply only to a received QSIG message that has been successfully validated and that satisfies one of the following conditions: -the QSIG message is a SETUP message and indicates a destination in the IP network and a bearer capability for which the gateway is able to provide interworking; or -the QSIG message is a message other than SETUP and contains a call reference that identifies an existing call for which the gateway is providing interworking between QSIG and SIP. The processing of any valid QSIG message that does not satisfy any of these conditions is outside the scope of this specification. Also, the processing of any QSIG message relating to call-independent signalling connections or connectionless transport, as specified in [3], is outside the scope of this specification. If segmented QSIG messages are received, the gateway SHALL await receipt of all segments of a message and SHALL validate and act on the complete reassembled message. The gateway SHALL validate received SIP messages (requests and responses) in accordance with the requirements of [10] and SHALL act in accordance with [10] on detection of a SIP protocol error.
Requirements of this section for acting on a received SIP message
apply only to a received message that has been successfully validated
and that satisfies one of the following conditions:
- the SIP message is an INVITE request that contains no tag parameter
in the To header field, does not match an ongoing transaction
(i.e., is not a merged request; see Section 8.2.2.2 of [10]), and
indicates a destination in the PISN for which the gateway is able
to provide interworking; or
- the SIP message is a request that relates to an existing dialog
representing a call for which the gateway is providing interworking
between QSIG and SIP; or
- the SIP message is a CANCEL request that relates to a received
INVITE request for which the gateway is providing interworking with
QSIG but for which the only response sent is informational (1xx),
no dialog having been confirmed; or
- the SIP message is a response to a request sent by the gateway in
accordance with this section.
The processing of any valid SIP message that does not satisfy any of
these conditions is outside the scope of this specification.
NOTE: These rules mean that an error detected in a received message
will not be propagated to the other side of the gateway. However,
there can be an indirect impact on the other side of the gateway,
e.g., the initiation of call clearing procedures.
The gateway SHALL run QSIG protocol timers as specified in [2] and
SHALL act in accordance with [2] if a QSIG protocol timer expires.
Any other action on expiry of a QSIG protocol timer is outside the
scope of this specification, except that if it results in the
clearing of the QSIG call, the gateway SHALL also clear the SIP call
in accordance with Section 8.4.5.
The gateway SHALL run SIP protocol timers as specified in [10] and
SHALL act in accordance with [10] if a SIP protocol timer expires.
Any other action on expiry of a SIP protocol timer is outside the
scope of this specification, except that if it results in the
clearing of the SIP call, the gateway SHALL also clear the QSIG call
in accordance with Section 8.4.5.
8.2. Call Establishment from QSIG to SIP
8.2.1. Call Establishment from QSIG to SIP Using En Bloc Procedures
The following procedures apply when the gateway receives a QSIG SETUP message containing a Sending Complete information element or the gateway receives a QSIG SETUP message and is able to determine that the number in the Called party number information element is complete. NOTE: In the absence of a Sending Complete information element, the means by which the gateway determines the number to be complete is an implementation matter. It can involve knowledge of the numbering plan and/or use of inter-digit timer expiry.8.2.1.1. Receipt of QSIG SETUP Message
On receipt of a QSIG SETUP message containing a number that the gateway determines to be complete in the Called party number information element, or containing a Sending complete information element and a number that could potentially be complete, the gateway SHALL map the QSIG SETUP message to a SIP INVITE request. The gateway SHALL also send a QSIG CALL PROCEEDING message. The gateway SHALL generate the SIP Request-URI, To, and From fields in the SIP INVITE request in accordance with Section 9. The gateway SHALL include in the INVITE request a Supported header containing option tag 100rel, to indicate support for [11]. The gateway SHALL include SDP offer information in the SIP INVITE request as described in Section 10. It SHOULD also connect the incoming media stream to the user information channel of the inter- PINX link, to allow the caller to hear in-band tones or announcements and prevent speech clipping on answer. Because of forking, the gateway may receive more than one media stream, in which case it SHOULD select one (e.g., the first received). If the gateway is able to correlate an unselected media stream with a particular early dialog established using a reliable provisional response, it MAY use the UPDATE method [19] to stop that stream and then use the UPDATE method to start that stream again if a 2xx response is received on that dialog. On receipt of a QSIG SETUP message containing a Sending complete information element and a number that the gateway determines to be incomplete in the Called party number information element, the gateway SHALL initiate QSIG call clearing procedures using cause value 28, "invalid number format (address incomplete)".
If information in the QSIG SETUP message is unsuitable for generating any of the mandatory fields in a SIP INVITE request (e.g., if a Request-URI cannot be derived from the QSIG Called party number information element) or for generating SDP information, the gateway SHALL NOT issue a SIP INVITE request and SHALL initiate QSIG call clearing procedures in accordance with [2].8.2.1.2. Receipt of SIP 100 (Trying) Response to an INVITE Request
A SIP 100 response SHALL NOT trigger any QSIG messages. It only serves the purpose of suppressing INVITE request retransmissions.8.2.1.3. Receipt of SIP 18x provisional response to an INVITE request
The gateway SHALL map a received SIP 18x response to an INVITE request to a QSIG PROGRESS or ALERTING message based on the following conditions. - If a SIP 180 response is received and no QSIG ALERTING message has been sent, the gateway SHALL generate a QSIG ALERTING message. The gateway MAY supply ring-back tone on the user information channel of the inter-PINX link, in which case the gateway SHALL include progress description number 8 in the QSIG ALERTING message. Otherwise the gateway SHALL NOT include progress description number 8 in the QSIG ALERTING message unless the gateway is aware that in-band information (e.g., ring-back tone) is being transmitted. - If a SIP 181/182/183 response is received, no QSIG ALERTING message has been sent, and no message containing progress description number 1 has been sent, the gateway SHALL generate a QSIG PROGRESS message containing progress description number 1. NOTE: This will ensure that QSIG timer T310 is stopped if running at the Originating PINX. In all other scenarios, the gateway SHALL NOT map the SIP 18x response to a QSIG message. If the SIP 18x response contains a Require header with option tag 100rel, the gateway SHALL send back a SIP PRACK request in accordance with [11].8.2.1.4. Receipt of SIP 2xx Response to an INVITE Request
If the gateway receives a SIP 2xx response as the first SIP 2xx response to a SIP INVITE request, the gateway SHALL map the SIP 2xx response to a QSIG CONNECT message. The gateway SHALL also send a SIP ACK request to acknowledge the 2xx response. The gateway SHALL
NOT include any SDP information in the SIP ACK request. If the gateway receives further 2xx responses, it SHALL respond to each in accordance with [10], SHOULD issue a BYE request for each, and SHALL NOT generate any further QSIG messages. Media streams will normally have been established in the IP network in each direction. If so, the gateway SHALL connect the media streams to the corresponding user-information channel on the inter- PINX link if it has not already done so and stop any local ring-back tone. If the SIP 2xx response is received in response to the SIP PRACK request, the gateway SHALL NOT map this message to any QSIG message. NOTE: A SIP 2xx response to the INVITE request can be received later on a different dialog as a result of a forking proxy.8.2.1.5. Receipt of SIP 3xx Response to an INVITE Request
On receipt of a SIP 3xx response to an INVITE request, the gateway SHALL act in accordance with [10]. NOTE: This will normally result in sending a new SIP INVITE request. Unless the gateway supports the QSIG Call Diversion Supplementary Service, no QSIG message SHALL be sent. The definition of Call Diversion Supplementary Service for QSIG to SIP interworking is beyond the scope of this specification.8.2.2. Call Establishment from QSIG to SIP Using Overlap Procedures
SIP uses en bloc signalling, and it is strongly RECOMMENDED to avoid using overlap signalling in a SIP network. A SIP/QSIG gateway dealing with overlap signalling SHOULD perform a conversion from overlap to en bloc signalling method using one or more of the following mechanisms: - timers; - numbering plan information; - the presence of a Sending complete information element in a received QSIG INFORMATION message. If the gateway performs a conversion from overlap to en bloc signalling in the SIP network, then the procedures defined in Section 8.2.2.1 SHALL apply.
However, for some applications it might be impossible to avoid using overlap signalling in the SIP network. In this case, the procedures defined in Section 8.2.2.2 SHALL apply.8.2.2.1. En Bloc Signalling in SIP Network
8.2.2.1.1. Receipt of QSIG SETUP Message
On receipt of a QSIG SETUP message containing no Sending complete information element and a number in the Called party number information element that the gateway cannot determine to be complete, the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start QSIG timer T302, and await further number digits.8.2.2.1.2. Receipt of QSIG INFORMATION Message
On receipt of each QSIG INFORMATION message containing no Sending complete information element and containing a number that the gateway cannot determine to be complete, QSIG timer T302 SHALL be restarted. When QSIG timer T302 expires or a QSIG INFORMATION message containing a Sending complete information element is received, the gateway SHALL send a SIP INVITE request as described in Section 8.2.1.1. The Request-URI and To fields (see Section 9) SHALL be generated from the concatenation of information in the Called party number information element in the received QSIG SETUP and INFORMATION messages. The gateway SHALL also send a QSIG CALL PROCEEDING message.8.2.2.1.3. Receipt of SIP Responses to INVITE Requests
SIP responses to INVITE requests SHALL be mapped as described in 8.2.1.8.2.2.2. Overlap Signalling in SIP Network
The procedures below for using overlap signalling in the SIP network are in accordance with the principles described in [18] for using overlap sending when interworking with ISDN User Part (ISUP). In [18], there is discussion of some potential problems arising from the use of overlap sending in the SIP network. These potential problems are applicable also in the context of QSIG-SIP interworking and can be avoided if overlap sending in the QSIG network is terminated at the gateway, in accordance with Section 8.2.2.1. The procedures below should be used only where it is not feasible to use the procedures of Section 8.2.2.1.
8.2.2.2.1. Receipt of QSIG SETUP Message
On receipt of a QSIG SETUP message containing no Sending complete information element and a number in the Called party number information element that the gateway cannot determine to be complete, the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message and start QSIG timer T302. If the QSIG SETUP message contains the minimum number of digits required to route the call in the IP network, the gateway SHALL send a SIP INVITE request as specified in Section 8.2.1.1. Otherwise, the gateway SHALL wait for more digits to arrive in QSIG INFORMATION messages.8.2.2.2.2. Receipt of QSIG INFORMATION Message
On receipt of a QSIG INFORMATION message, the gateway SHALL handle the QSIG timer T302 in accordance with [2]. NOTE: [2] requires the QSIG timer to be stopped if the INFORMATION message contains a Sending complete information element or to be restarted otherwise. Further behaviour of the gateway SHALL depend on whether or not it has already sent a SIP INVITE request. If the gateway has not sent a SIP INVITE request and it now has the minimum number of digits required to route the call, it SHALL send a SIP INVITE request as specified in Section 8.2.2.1.2. If the gateway still does not have the minimum number of digits required, it SHALL wait for more QSIG INFORMATION messages to arrive. If the gateway has already sent one or more SIP INVITE requests, whether or not final responses to those requests have been received, it SHALL send a new SIP INVITE request in accordance with Section 3.2 of [18]. The updated Request-URI and To fields (see Section 9) SHALL be generated from the concatenation of information in the Called party number information element in the received QSIG SETUP and INFORMATION messages. NOTE: [18] requires the new request to have the same Call-ID and the same From header (including tag) as in the previous INVITE request. [18] recommends that the CSeq header should contain a value higher than that in the previous INVITE request.8.2.2.2.3. Receipt of SIP 100 (Trying) Response to an INVITE Request
The requirements of Section 8.2.1.2 SHALL apply.
8.2.2.2.4. Receipt of SIP 18x Provisional Response to an INVITE Request
The requirements of Section 8.2.1.3 SHALL apply.8.2.2.2.5. Receipt of SIP 2xx Response to an INVITE Request
The requirements of Section 8.2.1.4 SHALL apply. In addition, the gateway SHALL send a SIP CANCEL request in accordance with Section 3.4 of [18] to cancel any SIP INVITE transactions for which no final response has been received.8.2.2.2.6. Receipt of SIP 3xx Response to an INVITE Request
The requirements of Section 8.2.1.5 SHALL apply.8.2.2.2.7. Receipt of a SIP 4xx, 5xx, or 6xx Final Response to an INVITE Request
On receipt of a SIP 4xx, 5xx, or 6xx final response to an INVITE request, the gateway SHALL send back a SIP ACK request. Unless the gateway is able to retry the INVITE request to avoid the problem (e.g., by supplying authentication in the case of a 401 or 407 response), the gateway SHALL also send a QSIG DISCONNECT message (8.4.4) if no further QSIG INFORMATION messages are expected and final responses have been received to all transmitted SIP INVITE requests. NOTE: Further QSIG INFORMATION messages will not be expected after QSIG timer T302 has expired or after a Sending complete information element has been received. In all other cases, the receipt of a SIP 4xx, 5xx, or 6xx final response to an INVITE request SHALL NOT trigger the sending of any QSIG message. NOTE: If further QSIG INFORMATION messages arrive, these will result in further SIP INVITE requests being sent, one of which might result in successful call establishment. For example, initial INVITE requests might produce 484 (Address Incomplete) or 404 (Not Found) responses because the Request-URIs derived from incomplete numbers cannot be routed, yet a subsequent INVITE request with a routable Request-URI might produce a 2xx final response or a more meaningful 4xx, 5xx, or 6xx final response.
8.2.2.2.8. Receipt of Multiple SIP Responses to an INVITE Request
Section 3.3 of [18] applies.8.2.2.2.9. Cancelling Pending SIP INVITE Transactions
As stated in Section 3.4 of [18], when a gateway sends a new SIP INVITE request containing new digits, it SHOULD NOT send a SIP CANCEL request to cancel a previous SIP INVITE transaction that has not had a final response. This SIP CANCEL request could arrive at an egress gateway before the new SIP INVITE request and trigger premature call clearing. NOTE: Previous SIP INVITE transactions can be expected to result in SIP 4xx class responses, which terminate the transaction. In Section 8.2.2.2.5, there is provision for cancelling any transactions still in progress after a SIP 2xx response has been received.8.2.2.2.10. QSIG Timer T302 Expiry
If QSIG timer T302 expires and the gateway has received 4xx, 5xx, or 6xx responses to all transmitted SIP INVITE requests, the gateway SHALL send a QSIG DISCONNECT message. If T302 expires and the gateway has not received 4xx, 5xx, or 6xx responses to all transmitted SIP INVITE requests, the gateway SHALL ignore any further QSIG INFORMATION messages but SHALL NOT send a QSIG DISCONNECT message at this stage. NOTE: A QSIG DISCONNECT request will be sent when all outstanding SIP INVITE requests have received 4xx, 5xx, or 6xx responses.8.3. Call Establishment from SIP to QSIG
8.3.1. Receipt of SIP INVITE Request for a New Call
On receipt of a SIP INVITE request for a new call, if a suitable channel is available on the inter-PINX link, the gateway SHALL generate a QSIG SETUP message from the received SIP INVITE request. The gateway SHALL generate the Called party number and Calling party number information elements in accordance with Section 9 and SHALL generate the Bearer capability information element in accordance with Section 10. If the gateway can determine that the number placed in the Called party number information element is complete, the gateway MAY include the Sending complete information element. NOTE: The means by which the gateway determines the number to be complete is an implementation matter. It can involve knowledge of the numbering plan and/or use of the inter-digit timer.
The gateway SHOULD send a SIP 100 (Trying) response. If information in the SIP INVITE request is unsuitable for generating any of the mandatory information elements in a QSIG SETUP message (e.g., if a QSIG Called party number information element cannot be derived from SIP Request-URI field) or if no suitable channel is available on the inter-PINX link, the gateway SHALL NOT issue a QSIG SETUP message and SHALL send a SIP 4xx, 5xx, or 6xx response. If no suitable channel is available, the gateway should use response code 503 (Service Unavailable). If the SIP INVITE request does not contain SDP information and does not contain either a Required header or a Supported header with option tag 100rel, the gateway SHOULD still proceed as above, although an implementation can instead send a SIP 488 (Not Acceptable Here) response, in which case it SHALL NOT issue a QSIG SETUP message. NOTE: The absence of SDP offer information in the SIP INVITE request means that the gateway might need to send SDP offer information in a provisional response and receive SDP answer information in a SIP PRACK request (in accordance with [11]) in order to ensure that tones and announcements from the PISN are transmitted. SDP offer information cannot be sent in an unreliable provisional response because SDP answer information would need to be returned in a SIP PRACK request. The recommendation above still to proceed with call establishment in this situation reflects the desire to maximise the chances of a successful call. However, if important in-band information is likely to be denied in this situation, a gateway can choose not to proceed. NOTE: If SDP offer information is present in the INVITE request, the issuing of a QSIG SETUP message is not dependent on the presence of a Required header or a Supported header with option tag 100rel. On receipt of a SIP INVITE request relating to a call that has already been established from SIP to QSIG, the procedures of 8.3.9 SHALL apply.8.3.2. Receipt of QSIG CALL PROCEEDING Message
The receipt of a QSIG CALL PROCEEDING message SHALL NOT result in any SIP message being sent.
8.3.3. Receipt of QSIG PROGRESS Message
A QSIG PROGRESS message can be received in the event of interworking on the remote side of the PISN or if the PISN is unable to complete the call and generates an in-band tone or announcement. In the latter case, a Cause information element is included in the QSIG PROGRESS message. The gateway SHALL map a received QSIG PROGRESS message to a SIP 183 (Session Progress) response to the INVITE request. If the SIP INVITE request contained either a Require header or a Supported header with option tag 100rel, the gateway SHALL include in the SIP 183 response a Require header with option tag 100rel. NOTE: In accordance with [11], inclusion of option tag 100rel in a provisional response instructs the UAC to acknowledge the provisional response by sending a PRACK request. [11] also specifies procedures for repeating a provisional response with option tag 100rel if no PRACK is received. If the QSIG PROGRESS message contained a Progress indicator information element with Progress description number 1 or 8, the gateway SHALL connect the media streams to the corresponding user information channel of the inter-PINX link if it has not already done so, provided that SDP answer information is included in the transmitted SIP response to the INVITE request or has already been sent or received. Inclusion of SDP offer or answer information in the 183 provisional response SHALL be in accordance with Section 8.3.5. If the QSIG PROGRESS message is received with a Cause information element, the gateway SHALL either wait until the tone/announcement is complete or has been applied for sufficient time before initiating call clearing, or wait for a SIP CANCEL request. If call clearing is initiated, the cause value in the QSIG PROGRESS message SHALL be used to derive the response to the SIP INVITE request in accordance with Table 1.8.3.4. Receipt of QSIG ALERTING Message
The gateway SHALL map a QSIG ALERTING message to a SIP 180 (Ringing) response to the INVITE request. If the SIP INVITE request contained either a Require header or a Supported header with option tag 100rel, the gateway SHALL include in the SIP 180 response a Require header with option tag 100rel.
NOTE: In accordance with [11], inclusion of option tag 100rel in a provisional response instructs the UAC to acknowledge the provisional response by sending a PRACK request. [11] also specifies procedures for repeating a provisional response with option tag 100rel if no PRACK is received. If the QSIG ALERTING message contained a Progress indicator information element with Progress description number 1 or 8, the gateway SHALL connect the media streams to the corresponding user information channel of the inter-PINX link if it has not already done so, provided that SDP answer information is included in the transmitted SIP response or has already been sent or received. Inclusion of SDP offer or answer information in the 180 provisional response SHALL be in accordance with Section 8.3.5.8.3.5. Inclusion of SDP Information in a SIP 18x Provisional Response
When sending a SIP 18x provisional response to the INVITE request, if a QSIG message containing a Progress indicator information element with progress description number 1 or 8 has been received the gateway SHALL include SDP information. Otherwise, the gateway MAY include SDP information. If SDP information is included, it shall be in accordance with the following rules. If the SIP INVITE request contained a Required or Supported header with option tag 100rel, and if SDP offer and answer information has already been exchanged, no SDP information SHALL be included in the SIP 18x provisional response. If the SIP INVITE request contained a Required or Supported header with option tag 100rel, and if SDP offer information was received in the SIP INVITE request but no SDP answer information has been sent, SDP answer information SHALL be included in the SIP 18x provisional response. If the SIP INVITE request contained a Required or Supported header with option tag 100rel, and if no SDP offer information was received in the SIP INVITE request and no SDP offer information has already been sent, SDP offer information SHALL be included in the SIP 18x provisional response. NOTE: In this case, SDP answer information can be expected in the SIP PRACK. If the SIP INVITE request contained neither a Required nor a Supported header with option tag 100rel, SDP answer information SHALL be included in the SIP 18x provisional response.
NOTE: Because the provisional response is unreliable, SDP answer information needs to be repeated in each provisional response and in the final SIP 2xx response. NOTE: If the SIP INVITE request contained no SDP offer information and neither a Required nor a Supported header with option tag 100rel, it should have been rejected in accordance with Section 8.3.1.8.3.6. Receipt of QSIG CONNECT Message
The gateway SHALL map a QSIG CONNECT message to a SIP 200 (OK) final response for the SIP INVITE request. The gateway SHALL also send a QSIG CONNECT ACKNOWLEDGE message. If the SIP INVITE request contained a Required or Supported header with option tag 100rel, and if SDP offer and answer information has already been exchanged, no SDP information SHALL be included in the SIP 200 response. If the SIP INVITE request contained a Required or Supported header with option tag 100rel, and if SDP offer information was received in the SIP INVITE request but no SDP answer information has been sent, SDP answer information SHALL be included in the SIP 200 response. If the SIP INVITE request contained a Required or Supported header with option tag 100rel, and if no SDP offer information was received in the SIP INVITE request and no SDP offer information has already been sent, SDP offer information SHALL be included in the SIP 200 response. NOTE: In this case, SDP answer information can be expected in the SIP ACK. If the SIP INVITE request contained neither a Required nor a Supported header with option tag 100rel, SDP answer information SHALL be included in the SIP 200 response. NOTE: Because the provisional response is unreliable, SDP answer information needs to be repeated in each provisional response and in the final 2xx response. NOTE: If the SIP INVITE request contained no SDP offer information and neither a Required nor a Supported header with option tag 100rel, it may have been rejected in accordance with Section 8.3.1.
The gateway SHALL connect the media streams to the corresponding user information channel of the inter-PINX link if it has not already done so, provided that SDP answer information is included in the transmitted SIP response or has already been sent or received.8.3.7. Receipt of SIP PRACK Request
The receipt of a SIP PRACK request acknowledging a reliable provisional response SHALL NOT result in any QSIG message being sent. The gateway SHALL send back a SIP 200 (OK) response to the SIP PRACK request. If the SIP PRACK contains SDP answer information and a QSIG message containing a Progress indicator information element with progress description number 1 or 8 has been received, the gateway SHALL connect the media streams to the corresponding user information channel of the inter-PINX link.8.3.8. Receipt of SIP ACK Request
The receipt of a SIP ACK request SHALL NOT result in any QSIG message being sent. If the SIP ACK contains SDP answer information, the gateway SHALL connect the media streams to the corresponding user information channel of the inter-PINX link if it has not already done so.8.3.9. Receipt of a SIP INVITE Request for a Call Already Being Established
A gateway can receive a call from SIP using overlap procedures. This should occur when the UAC for the INVITE request is a gateway from a network that employs overlap procedures (e.g., an ISUP gateway or another QSIG gateway) and the gateway has not absorbed overlap. For a call from SIP using overlap procedures, the gateway will receive multiple SIP INVITE requests that belong to the same call but have different Request-URI and To fields. Each SIP INVITE request belongs to a different dialog. A SIP INVITE request is considered to be for the purpose of overlap sending if, compared to a previously received SIP INVITE request, it has: - the same Call-ID header; - the same From header (including the tag); - no tag in the To header;
- an updated Request-URI from which can be derived a called party
number with a superset of the digits derived from the previously
received SIP INVITE request;
and if
- the gateway has not yet sent a final response other than 484 to
the previously received SIP INVITE request.
If a gateway receives a SIP INVITE request for the purpose of overlap
sending, it SHALL generate a QSIG INFORMATION message using the call
reference of the existing QSIG call instead of a new QSIG SETUP
message and containing only the additional digits in the Called party
number information element. It SHALL also respond to the SIP INVITE
request received previously with a SIP 484 Address Incomplete
response.
If a gateway receives a SIP INVITE request that meets all of the
conditions for a SIP INVITE request for the purpose of overlap
sending except the condition concerning the Request-URI, the gateway
SHALL respond to the new request with a SIP 485 (Ambiguous) response.
8.4. Call Clearing and Call Failure
8.4.1. Receipt of a QSIG DISCONNECT, RELEASE, or RELEASE COMPLETE
Message
On receipt of QSIG DISCONNECT, RELEASE, or RELEASE COMPLETE message
as the first QSIG call clearing message, gateway behaviour SHALL
depend on the state of call establishment.
1) If the gateway has sent a SIP 200 (OK) response to a SIP INVITE
request and received a SIP ACK request, or if it has received a
SIP 200 (OK) response to a SIP INVITE request and sent a SIP ACK
request, the gateway SHALL send a SIP BYE request to clear the
call.
2) If the gateway has sent a SIP 200 (OK) response to a SIP INVITE
request (indicating that call establishment is complete) but has
not received a SIP ACK request, the gateway SHALL wait until a SIP
ACK is received and then send a SIP BYE request to clear the call.
3) If the gateway has sent a SIP INVITE request and received a SIP
provisional response but not a SIP final response, the gateway
SHALL send a SIP CANCEL request to clear the call.
NOTE 1: In accordance with [10], if after sending a SIP CANCEL
request a SIP 2xx response is received to the SIP INVITE request,
the gateway will need to send a SIP BYE request.
4) If the gateway has sent a SIP INVITE request but received no SIP
response, the gateway SHALL NOT send a SIP message. If a SIP
final or provisional response is subsequently received, the
gateway SHALL then act in accordance with 1, 2, or 3 above,
respectively.
5) If the gateway has received a SIP INVITE request but not sent a
SIP final response, the gateway SHALL send a SIP final response
chosen according to the cause value in the received QSIG message
as specified in Table 1. SIP response 500 (Server internal error)
SHALL be used as the default for cause values not shown in
Table 1.
NOTE 2: It is not necessarily appropriate to map some QSIG cause
values to SIP messages because these cause values are meaningful only
at the gateway. A good example of this is cause value 44, "Requested
circuit or channel not available", which signifies that the channel
number in the transmitted QSIG SETUP message was not acceptable to
the peer PINX. The appropriate behavior in this case is for the
gateway to send another SETUP message indicating a different channel
number. If this is not possible, the gateway should treat it either
as a congestion situation (no channels available; see Section 8.3.1)
or as a gateway failure situation (in which case the default SIP
response code applies).
In all cases, the gateway SHALL also disconnect media streams, if
established, and allow QSIG and SIP signalling to complete in
accordance with [2] and [10], respectively.
Table 1: Mapping of QSIG Cause Value to SIP 4xx-6xx responses to an
INVITE request
QSIG Cause value SIP response
----------------------------------------------------------------
1 Unallocated number 404 Not found
2 No route to specified 404 Not found
transit network
3 No route to destination 404 Not found
16 Normal call clearing (NOTE 3)
17 User busy 486 Busy here
18 No user responding 408 Request timeout
19 No answer from the user 480 Temporarily unavailable
20 Subscriber absent 480 Temporarily unavailable
21 Call rejected 603 Decline, if location field
in Cause information element
indicates user. Otherwise:
403 Forbidden
22 Number changed 301 Moved permanently, if
information in diagnostic field
of Cause information element is
suitable for generating a SIP
Contact header. Otherwise:
410 Gone
23 Redirection to new 410 Gone
destination
27 Destination out of order 502 Bad gateway
28 Address incomplete 484 Address incomplete
29 Facility rejected 501 Not implemented
31 Normal, unspecified 480 Temporarily unavailable
34 No circuit/channel 503 Service unavailable
available
38 Network out of order 503 Service unavailable
41 Temporary failure 503 Service unavailable
42 Switching equipment 503 Service unavailable
congestion
47 Resource unavailable, 503 Service unavailable
unspecified
55 Incoming calls barred 403 Forbidden
within CUG
57 Bearer capability not 403 Forbidden
authorized
58 Bearer capability not 503 Service unavailable
presently available
65 Bearer capability not 488 Not acceptable here (NOTE 4)
implemented
69 Requested facility not 501 Not implemented
implemented
70 Only restricted digital 488 Not acceptable here (NOTE 4)
information available
79 Service or option not 501 Not implemented
implemented, unspecified
87 User not member of CUG 403 Forbidden
88 Incompatible destination 503 Service unavailable
102 Recovery on timer expiry 504 Server time-out
NOTE 3: A QSIG call clearing message containing cause value 16 will
normally result in the sending of a SIP BYE or CANCEL request.
However, if a SIP response is to be sent to the INVITE request, the
default response code should be used.
NOTE 4: The gateway may include a SIP Warning header if diagnostic
information in the QSIG Cause information element allows a suitable
warning code to be selected.
8.4.2. Receipt of a SIP BYE Request
On receipt of a SIP BYE request, the gateway SHALL send a QSIG
DISCONNECT message with cause value 16 (normal call clearing). The
gateway SHALL also disconnect media streams, if established, and
allow QSIG and SIP signalling to complete in accordance with [2] and
[10], respectively.
NOTE: When responding to a SIP BYE request, in accordance with [10],
the gateway is also required to respond to any other outstanding
transactions, e.g., with a SIP 487 (Request Terminated) response.
This applies in particular if the gateway has not yet returned a
final response to the SIP INVITE request.
8.4.3. Receipt of a SIP CANCEL Request
On receipt of a SIP CANCEL request to clear a call for which the
gateway has not sent a SIP final response to the received SIP INVITE
request, the gateway SHALL send a QSIG DISCONNECT message with cause
value 16 (normal call clearing). The gateway SHALL also disconnect
media streams, if established, and allow QSIG and SIP signalling to
complete in accordance with [2] and [10], respectively.
8.4.4. Receipt of a SIP 4xx-6xx Response to an INVITE Request
Except where otherwise specified in the context of overlap sending
(8.2.2.2), on receipt of a SIP final response (4xx-6xx) to a SIP
INVITE request, unless the gateway is able to retry the INVITE
request to avoid the problem (e.g., by supplying authentication in
the case of a 401 or 407 response), the gateway SHALL transmit a QSIG
DISCONNECT message. The cause value in the QSIG DISCONNECT message
SHALL be derived from the SIP 4xx-6xx response according to Table 2. Cause value 31 (Normal, unspecified) SHALL be used as the default for SIP responses not shown in Table 2. The gateway SHALL also disconnect media streams, if established, and allow QSIG and SIP signalling to complete in accordance with [2] and [10], respectively. When generating a QSIG Cause information element, the location field SHOULD contain the value "user", if generated as a result of a SIP response code 6xx, or the value "Private network serving the remote user" in other circumstances. Table 2: Mapping of SIP 4xx-6xx responses to an INVITE request to QSIG Cause values SIP response QSIG Cause value (NOTE 6) ------------------------------------------------------------------ 400 Bad request 41 Temporary failure 401 Unauthorized 21 Call rejected (NOTE 5) 402 Payment required 21 Call rejected 403 Forbidden 21 Call rejected 404 Not found 1 Unallocated number 405 Method not allowed 63 Service or option unavailable, unspecified 406 Not acceptable 79 Service or option not implemented, unspecified 407 Proxy Authentication required 21 Call rejected (NOTE 5) 408 Request timeout 102 Recovery on timer expiry 410 Gone 22 Number changed 413 Request entity too large 127 Interworking, unspecified (NOTE 6) 414 Request-URI too long 127 Interworking, unspecified (NOTE 6) 415 Unsupported media type 79 Service or option not implemented, unspecified (NOTE 6) 416 Unsupported URI scheme 127 Interworking, unspecified (NOTE 6) 420 Bad extension 127 Interworking, unspecified (NOTE 6) 421 Extension required 127 Interworking, unspecified (NOTE 6) 423 Interval too brief 127 Interworking, unspecified (NOTE 6) 480 Temporarily unavailable 18 No user responding 481 Call/transaction does not exist 41 Temporary failure 482 Loop detected 25 Exchange routing error 483 Too many hops 25 Exchange routing error
484 Address incomplete 28 Invalid number format
(NOTE 6)
485 Ambiguous 1 Unallocated Number
486 Busy here 17 User busy
487 Request terminated (NOTE 7)
488 Not Acceptable Here 65 Bearer capability not
implemented or 31 Normal,
unspecified (NOTE 8)
500 Server internal error 41 Temporary failure
501 Not implemented 79 Service or option not
implemented, unspecified
502 Bad gateway 38 Network out of order
503 Service unavailable 41 Temporary failure
504 Gateway time-out 102 Recovery on timer expiry
505 Version not supported 127 Interworking, unspecified
(NOTE 6)
513 Message too large 127 Interworking, unspecified
(NOTE 6)
600 Busy everywhere 17 User busy
603 Decline 21 Call rejected
604 Does not exist anywhere 1 Unallocated number
606 Not acceptable 65 Bearer capability not
implemented or
31 Normal, unspecified (NOTE 8)
NOTE 5: In some cases, it may be possible for the gateway to provide
credentials to the SIP UAS that is rejecting an INVITE due to
authorization failure. If the gateway can authenticate itself, then
obviously it should do so and proceed with the call. Only if the
gateway cannot authorize itself should the gateway clear the call in
the QSIG network with this cause value.
NOTE 6: For some response codes, the gateway may be able to retry the
INVITE request in order to work around the problem. In particular,
this may be the case with response codes indicating a protocol error.
The gateway SHOULD clear the call in the QSIG network with the
indicated cause value only if retry is not possible or fails.
NOTE 7: The circumstances in which SIP response code 487 can be
expected to arise do not require it to be mapped to a QSIG cause
code, since the QSIG call will normally already be cleared or in the
process of clearing. If QSIG call clearing does, however, need to be
initiated, the default cause value should be used.
NOTE 8: When the Warning header is present in a SIP 606 or 488
message, the warning code should be examined to determine whether it
is reasonable to generate cause value 65. This cause value should be
generated only if there is a chance that a new call attempt with
different content in the Bearer capability information element will avoid the problem. In other circumstances, the default cause value should be used.8.4.5 Gateway-Initiated Call Clearing
If the gateway initiates clearing of the QSIG call owing to QSIG timer expiry, QSIG protocol error, or use of the QSIG RESTART message in accordance with [2], the gateway SHALL also initiate clearing of the SIP call in accordance with Section 8.4.1. If this involves the sending of a final response to a SIP INVITE request, the gateway SHALL use response code 480 (Temporarily Unavailable) if optional QSIG timer T301 has expired or, otherwise, response code 408 (Request timeout) or 500 (Server internal error), as appropriate. If the gateway initiates clearing of the SIP call owing to SIP timer expiry or SIP protocol error in accordance with [10], the gateway SHALL also initiate clearing of the QSIG call in accordance with [2] using cause value 102 (Recovery on timer expiry) or 41 (Temporary failure), as appropriate.8.5. Request to Change Media Characteristics
If after a call has been successfully established the gateway receives a SIP INVITE request to change the media characteristics of the call in a way that would be incompatible with the bearer capability in use within the PISN, the gateway SHALL send back a SIP 488 (Not Acceptable Here) response and SHALL NOT change the media characteristics of the existing call.
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