📄 rfc3087.txt
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could be chosen if the subscriber's contact list has been otherwise
exhausted with no answer. The busy-announcement URI would be chosen
when a busy everywhere response is received from one of the contacts.
A DND announcement URI could be selected if the subscriber had
activated DND. Calls to sip:receptionist@wcom.com could be configured
to roll to sip:deposit@wcom.com
Campbell & Sparks Informational [Page 6]
RFC 3087 SIP Service Control April 2001
3.2 Retrievals
3.2.1 Request to Retrieve from a particular mailbox
3.2.1.1 Trusted SIP source
A request to retrieve the contents of a particular mailbox (sip:sub-
rjs-retrieve@vm.wcom.com) coming from a trusted source could be
honored without further authentication checks. A trusted source is
one with which the voice mail service has secure communications, and
to which it is willing to delegate authentication. This could be the
service's protecting proxy for example.
3.2.1.2 Authenticated SIP source
A service, or its protecting proxy, could choose to honor a retrieve
request for a particular mailbox (sip:sub-rjs-retrieve@vm.wcom.com)
based on SIP authentication. If SIP level authentication failed, the
service or proxy could be configured to send the call to the in-band
pin prompting URI (sip:sub-rjs-retrieve-inpin@vm.wcom.com).
3.2.1.3 Unauthenticated SIP source
A service, or its protecting proxy, receiving a retrieve request for
a particular mailbox (sip:sub-rjs-retrieve@vm.wcom.com) with no other
method of authenticating the requestor could send the request to the
in-band pin prompting URI (sip:sub-rjs-retrieve-inpin@vm.wcom.com).
3.2.1.4 PSTN source
This scenario assumes that the service provider's network has been
configured such that a PSTN number could be dialed explicitly for
retrieving messages from a particular mailbox. Such services
currently exist, but are not common. In such a network, the
gateway's proxy would map the call to the in-band pin prompting URI
(sip:sub-rjs-retrieve-inpin@vm.wcom.com).
3.2.2 Request to Retrieve, mailbox to be determined
3.2.2.1 SIP source
As in the Request to Deposit scenario, when a service receives a
request for the top level retrieve URI it would most likely need to
use in-band IVR techniques to determine the target mailbox and
authenticate the caller.
Campbell & Sparks Informational [Page 7]
RFC 3087 SIP Service Control April 2001
3.2.2.2 Arbitrary PSTN source
This scenario assumes there is a single PSTN number that subscribers
dial to access the voice mail service to retrieve messages. This is
the most common access method provided by current voice mail
services.
The gateway's proxy would map a call to the top level PSTN number to
the top level retrieve in-band prompting URI (sip:retrieve-
in@vm.wcom.com). Once the system identifies the target mailbox, the
call would be transferred to the appropriate in-band pin prompting
URI (sip:sub-rjs-retrieve-inpin@vm.wcom.com).
3.2.2.3 Recognized PSTN source
This scenario also assumes there is a single PSTN number that
subscribers dial to access the voice mail service to retrieve
messages.
The gateway's proxy would recognize the calling party number as a
subscriber, and map the call to the subscriber's in-band prompting
URI (sip:sub-rjs-retrieve-inpin@vm.wcom.com)
4. Voice Mail Call Flow Examples
The following section describes some example call flows for a
hypothetical voice mail service, with the host name of vm.wcom.com.
All the call flows assume that a proxy protects the voice mail
service and that a trust relationship exists between the voice mail
service and the proxy.
4.1 Generic Scenario
4.1.1 Direct call to the voice mail system
User A calls the voice mail system directly. The voice mail system
invokes the top-level menu, which might prompt the caller for an
extension or the first few letters of a name.
Campbell & Sparks Informational [Page 8]
RFC 3087 SIP Service Control April 2001
User A Proxy VM Service
| | |
| INVITE F1 | |
|------------------>| |
| | INVITE F2 |
| (100 Trying) F3 |---------------------->|
|<------------------| |
| | 180 Ringing F4 |
| 180 Ringing F5 |<----------------------|
|<------------------| |
| | 200 OK F6 |
| 200 OK F6 |<----------------------|
|<------------------| |
| | |
| ACK F8 | |
|------------------>| ACK F9 |
| |---------------------->|
| | |
| RTP Established- Play top level menu |
|<-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m-m->|
| | |
| BYE F10 | |
|------------------>| BYE F11 |
| |---------------------->|
| | |
| | 200 OK F12 |
| |<----------------------|
| 200 OK F13 | |
|<------------------| |
| | |
Flow Id Comments
INVITE F1 INVITE sip:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Proxy-Authorization:Digest username="UserA",
realm="MCI WorldCom SIP",
nonce="ea9c8e88df84f1cc4e341ae6cbe5a359", opaque="",
uri="sip:VoiceMail@wcom.com", response=<appropriately
calculated hash goes here>
Content-Type: application/sdp
Content-Length: <appropriate value>
Campbell & Sparks Informational [Page 9]
RFC 3087 SIP Service Control April 2001
v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/*Client for A prepares to receive data on port 49170
from the network. */
INVITE F2 INVITE sip:top@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: TheBigGuy <sip:UserA@here.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
v=0
o=UserA 2890844526 2890844526 IN IP4 client.here.com
s=Session SDP
c=IN IP4 100.101.102.103
t=0 0
m=audio 49170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
(100 Trying SIP/2.0 100 Trying
F3 Via: SIP/2.0/UDP here.com:5060
Proxy->A) From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
180 Ringing SIP/2.0 180 Ringing
F4 Via: SIP/2.0/UDP wcom.com:5060; branch=1
VM->Proxy Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
Campbell & Sparks Informational [Page 10]
RFC 3087 SIP Service Control April 2001
180 Ringing SIP/2.0 180 Ringing
F5 Via: SIP/2.0/UDP here.com:5060
Proxy->A From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Content-Length: 0
200 OK F6 SIP/2.0 200 OK
VM->Proxy Via: SIP/2.0/UDP wcom.com:5060; branch=1
Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact: VoiceMailSystem <sip:top@vm.wcom.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
200 OK F7 SIP/2.0 200 OK
Proxy->A Via: SIP/2.0/UDP here.com:5060
Record-Route: <sip:VoiceMail@wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 INVITE
Contact VoiceMailSystem <sip:top@vm.wcom.com>
Content-Type: application/sdp
Content-Length: <appropriate value>
v=0
o=UserB 2890844527 2890844527 IN IP4 vm.wcom.com
s=Session SDP
c=IN IP4 110.111.112.114
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Campbell & Sparks Informational [Page 11]
RFC 3087 SIP Service Control April 2001
ACK F8 ACK sip:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip:top@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
ACK F9 ACK sip:top@vm.wcom.com SIP/2.0
Proxy->VM Via: SIP/2.0/UDP wcom.com:5060
Via: SIP/2.0/UDP here.com:5060
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>; tag=3145678
Call-Id: 12345600@here.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between A and VM. VM
system starts IVR dialog for top level menu */
/* User A Hangs Up with VM system. Alternatively, the
VM system could initiate the BYE*/
BYE F10 BYE sip:VoiceMail@wcom.com SIP/2.0
A->Proxy Via: SIP/2.0/UDP here.com:5060
Route:<sip: top@vm.wcom.com>
From: TheBigGuy <sip:UserA@here.com>
To: VoiceMail <sip:VoiceMail@wcom.com>;tag=3145678
Call-Id: 12345600@here.com
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