Cisco SLT Example [View full size image] NOTE For additional information about Cisco Softswitch products, including the PGW2200 and BTS10200, visit the following Web site: http://www.cis
Trang 1ISDN User Adaptation (IUA)
In addition to addressing SS7 over IP, the SigTran group also addressed the
backhaul of ISDN over an IP network RFC 3057 [142] defined the IUA, which is supplemented by an Implementer's Guide [143] that seamlessly supports the Q.921 user (Q.931 and QSIG) It also supports both ISDN Primary Rate Access (PRA) and Basic Rate Access (BRA) as well as Facility Associated Signaling (FAS), Non-Facility Associated Signaling (NFAS), and NFAS with backup D channel Further, extensions to IUA are defined for DPNSS/DASS2 [144], V5.2 [145], and
GR 303 [146] that will most likely become RFCs in the future
Figure 14-25
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Trang 2Early Cisco SS7/IP Solution
Cisco was working on a SLT device before the SS7/IP IETF standardization efforts began The Cisco SLT is a modular access router (Cisco 2611 or 2651) that
terminates SS7 signaling links and backhauls MTP Level 3 and above to a PGW
2200 (formerly SC 2200 and VSC 3000) MGC Figure 14-26 shows an example configuration of two Cisco SLTs providing SS7 termination and backhaul for the Cisco PGW 2200 Softswitch
Figure 14-26 Cisco SLT Example
[View full size image]
NOTE
For additional information about Cisco Softswitch products, including the
PGW2200 and BTS10200, visit the following Web site:
http://www.cisco.com/en/US/products/sw/voicesw/index.html
The SLT supports either SS7 A-link or F-link configurations As noted previously, some SS7 links are deployed with bearer channels that are provisioned on the time slots that are not used by signaling channels The SLT supports a drop-and-insert feature, which allows the signaling channels to be groomed from the facility The bearer channels are hair pinnned on the interface card that is to be sent to a MG Figure 14-27 shows an example of the drop-and-insert feature
Figure 14-27 Example of SLT Drop-and-Insert Feature
[View full size image]
Each 2611 SLT can terminate up to two SS7 links, and the 2651 SLT can terminate
Trang 3up to four links Both have support for ANSI, ITU, TTC, and NTT variants
Several physical layer interfaces are supported on the SLT, including V.35, T1, and E1
The SLT function can also be integrated into the MG, as is done on some of the Cisco universal gateways The following Web site contains more information about the Cisco SLT:
http://www.cisco.com/en/US/products/hw/vcallcon/ps2152/products_data_sheet09 186a0080091b58.html
To deliver the backhauled messages to the PGW2200 reliably, the SLT makes use
of Reliable UDP (RUDP) and Session Manager (SM) protocols A generic
backhaul protocol layer is used to provide adaptation between MTP Level 2 and MTP Level 3 Figure 14-28 shows the protocol stacks used by the SLT and
PGW2200
Figure 14-28 Cisco SLT Protocol Stack
RUDP is a simple packet-based transport protocol that is based on Reliable Data Protocol (RFC 1151 [148] and RFC 908 [149]) RUDP has the following features:
• Connection-oriented
• Guarantees packet delivery with retransmission
• Maintains session connectivity using keepalive messages
• Provides notification of session failure
The SLT maintains up to two RUDP sessions to each PGW2200 host The use of two sessions provides for additional reliability because they provide for two
different network paths between the SLT and the PGW2200
The SM layer manages the RUDP sessions under control of the PGW2200 A single RUDP session is used to pass messages between the SLT and PGW2200 based on RUDP session availability and the PGW2200 hosts' Active/Standby state The Active PGW2200 selects one or two possible RUDP sessions and indicates its selection to the SLT via the SM protocol
Trang 4The generic backhaul protocol layer is very similar to M2UA; it provides the same basic functionality for backhauling MTP Level 3 and above over IP to the
PGW2200
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Trang 5SS7 and SIP/H.323 Interworking
The ITU-T originally developed the H.323 [125] for multimedia over Local Area Networks (LANs) It is not a single protocol; rather, it is a vertically-integrated suite of protocols that define the components and signaling Though it was
originally used for video-conferencing, H.323 was enhanced to better support VoIP with the Version 2 release It is currently the most widely-deployed VoIP solution today
One of the main complaints about H.323 is its complexity With H.323, many messages must be passed to set up even a basic voice call SIP [124], is considered
a simpler, more flexible alternative to H.323 SIP is a signaling protocol that
handles the setup, modification and teardown of multimedia sessions It was
developed in the IETF as a signaling protocol for establishing sessions in an IP network A session can be a simple two-way telephone call or a multimedia
conference SIP is becoming a popular favorite as the future of VoIP
So, how does SigTran play a role in H.323 and SIP? SigTran can provide PSTN connectivity to H.323 and SIP networks A PSTN Gateway application can be used
to fulfill this need The PSTN Gateway sits on the edge of the circuit-switched and packet-switched networks and provides SIP or H.323 interworking to SS7 in the PSTN Figure 14-29 shows an example of an SIP PSTN Gateway application In this example, the MGC connects to the SGs using SigTran
Figure 14-29 SIP-PSTN Gateway Application
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Figure 14-30 shows a similar example of an H.323 PSTN Gateway application
Figure 14-30 H.323-PSTN Gateway Application
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Trang 6Another interesting application is the PSTN transit application, in which calls originate and terminate on TDM interfaces and then transit a voice packet network (such as SIP or H.323) Service providers can use this application to offload their tandem and transit Class 4 and Class 3 switches This application creates the need for an ISUP transparency SIP-T [150] (SIP for Telephones) provides a framework for the integration of the PSTN with SIP Figure 14-31 shows an example of using SIP-T for a PSTN transit application
Figure 14-31 SIP Transit Application
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SIP-T meets the SS7 to SIP interworking requirements by providing the following functions:
• A standard way of mapping ISUP information into the SIP header for calls that originate in the PSTN This function ensures that the SIP contains
sufficient information to route calls (for example, in the case where routing depends on some ISUP information)
• Use of the SIP INFO [151] Method to transfer mid-call ISUP signaling messages
• A means for MIME [152] encapsulation of the ISUP signaling information
in the SIP body provides for ISUP transparency
When the MGC receives an ISUP message, the appropriate ISUP parameters are translated to the SIP header fields and the ISUP message is encapsulated in a MIME attachment, which intermediate SIP entities treat as an opaque object If the SIP message terminates the call, it ignores the ISUP attachment because it has no need for it However, if the call terminates on the PSTN, the encapsulated ISUP message is examined and used to generate the outgoing ISUP message The
version parameter included in the MIME media type information indicates the encapsulated ISUP message's ISUP variant If there are different ISUP variants on the origination and termination side, it is up to the terminating MGC to perform ISUP translation between the variants