Polycom Network Card Vega Gateway User Manual

Vega Gateway  
Scenarios  
A Pre-sales Engineer’s  
guide  
This guide has been written to assist the pre-sales engineers of VegaStream’s distributors and  
their resellers.  
The guide details a set of standard configurations along with key considerations and  
recommended 3rd party interoperable products.  
A guide to the scenarios and information you will find within this document are listed below.  
1. VoIP between sites  
2. VoIP between sites and home-worker / remote office  
3. VoIP between sites and home-worker / remote office using a proxy  
4. Analog long line extensions  
5. IP Telephony  
6. PSTN toll bypass  
7. TDM to / from VoIP converter  
8. Leased line eliminator for QSIG connections between sites  
9. Analog connections  
- Analog Applications  
a) Vega FXS for analog telephones  
b) Analog breakout to the PSTN – e.g. for a company or for a PSTN toll bypass  
c) FXS connection to a PBX  
d) FXO connection to a PBX  
e) PBX to PBX connectivity – using both FXS and FXO  
f) PBX to PBX connectivity – using FXS only  
g) PBX to PBX connectivity – using FXO only  
h) Mass analog connection – using a Vega 400 and a channel bank  
i) Analog long line extensions  
Appendix 1 - 3rd Party Products  
Appendix 2 - Things to consider  
Version 2.1  
16 March 2006  
Page 1 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
Telephony Network  
E1 Euro ISDN signalling is supported by Vega 400 and Vega 100  
T1 NI1 & NI2 signalling is supported by Vega 400 and Vega 100  
T1 Loop start, Ground start and Wink start CAS signalling is supported by Vega 400 and  
Vega 100  
BRI Euro ISDN signalling is supported by Vega 50 BRI  
Analog connection is possible, but is not preferred (for details on analog connectivity see  
later section)  
IP Network  
Recommend – private IP network between sites  
Can be any reliable IP connection, including leased line, line of site laser / wireless  
connection and satellite  
QOS should be implemented to ensure that VoIP packets pass through the network in a  
consistent and timely manner and are not discarded by congested routers  
Consider VPN between sites as this:  
- removes any issues with firewalls / NAT  
- encrypts audio and management data  
The Internet can be used, but contention and available bandwidth must be considered  
User experience  
By grooming the traffic on the PSTN side of the PBX all user PBX functions will continue to  
operate. This system will operate in exactly the same way as the phone system worked  
before the gateways were installed, for example, even existing shortcode dialling will  
continue to work (as the PBX will expand this to a full number before presenting it to the  
PSTN).  
USPs  
Vega dial planner capability provides powerful routing decision making. For example, calls  
can be routed based on dialled number prefixes, full numbers, and even on who is making  
the call.  
Considerations  
Check the numbers that are presented to the PSTN consider for example:  
- does the PBX present both National format and local format numbers to the PBX?  
- does the PBX insert any “carrier select prefix” to the dialled number?  
- can “block caller ID” or other prefixes be used in dialled numbers?  
Emergency calls should be routed directly to the PSTN, not over the VoIP link  
Program in fixed number length dial plans where possible to avoid the DTMF timeout  
delays incurred when routing calls to variable length numbers  
Call transfer from PBX to PBX will only work if call transfer PBX to PBX worked across the  
PSTN (the PBX may block this).  
If extension number dialling of the far end PBX is to be used (rather than just grooming off  
the full dialled destination number), remember the extension number will have to be  
prefixed with the PBX’s ‘connect to PSTN’ digit; usually 9 in UK, but can be 0 in Europe.  
If IP connectivity fails, calls can be programmed to be routed over the PSTN instead; there  
may, however, be a bit of delay in deciding that the IP connection is down.  
Version 2.1  
16 March 2006  
Page 3 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
Future extensions  
Add home-workers  
Extend the “on net” group to a group of companies who regularly call each other so all  
calls between these companies are free  
Connection to an ITSP for cheaper off-net calls – and possibly for more flexible ingress of  
national, international or non-geographic numbers.  
If PSTN connectivity is not reliable at any site then VoIP can be used as a backup to route  
outbound calls via a different site.  
More than 2 sites to be interconnected  
3rd party product choices  
For NAT traversal between sites, consider SNOM NAT filter, SIParator or Alcatel T610  
Version 2.1  
16 March 2006  
Page 4 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
2. VoIP between sites and home-worker / remote office  
Aim: To reduce inter-site call costs without changing the user experience, and adding in basic  
home-worker support  
Many multi-site companies find that staff making calls between sites is a significant company  
expense. Finding a method of reducing to zero the ongoing costs of these calls would be a real  
benefit to the bottom line. Also, with more and more people working from home – either  
permanently or just occasionally – it is useful and cost effective to have home-workers able to  
make free calls to office based staff and use the company PSTN connection for outbound  
company calls (no more telephony expenses forms, and appropriate tariffs for outbound calls  
based on those negotiated by the company with their chosen carrier).  
By inserting a Vega gateway between the PSTN and the PBX the Vega gateway can groom calls  
that are destined for other site(s) onto the IP network. All other calls from the PBX continue to be  
routed to the PSTN. All calls from the PSTN are routed to the PBX. The source Vega gateway will  
direct the VoIP calls to specific destination gateways, based on the telephone number dialled  
(single numbers or number ranges can be groomed off to specific destinations). On receiving a  
VoIP call, a destination gateway can use the dialled number to decide whether to route the call to  
the PBX of that site, or break out into the PSTN at that site (this is especially useful where  
different sites are in different countries – international toll call costs can be reduced to local or  
national call rates).  
If a significant proportion of the company’s calls are made between sites then it is possible that  
further cost savings can be made by reducing the number of channels connected to the PSTN.  
Version 2.1  
16 March 2006  
Page 5 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
Telephony Network  
- For PBXs  
E1 Euro ISDN signalling is supported by Vega 400 and Vega 100  
T1 NI1 & NI2 signalling is supported by Vega 400 and Vega 100  
T1 Loop start, Ground start and Wink start CAS signalling is supported by Vega 400 and  
Vega 100  
BRI Euro ISDN signalling is supported by Vega 50 BRI  
Analog connection is possible, but is not preferred (for details on analog connectivity see  
later section)  
- For home-workers  
Analog FXS gateway to connect to analog phone  
Optional FXO port to allow local connection to PSTN  
Alternatively use an IP handset or soft phone  
IP Network  
Recommend – private IP network between sites  
Can be any reliable IP connection, including leased line, line of site laser / wireless  
connection and satellite  
QOS should be implemented to ensure that VoIP packets pass through the network in a  
consistent and timely manner and are not discarded by congested routers  
Consider VPN between sites as this:  
- removes any issues with firewalls / NAT  
- encrypts audio and management data  
For inter-site operation the Internet can be used, but contention and available bandwidth  
must be considered  
Recommend – a VPN should be used from the home user to the company LAN. This is  
important for securing data access to the company site, and for VoIP it gets around NAT  
traversal issues  
User experience  
- For PBX users  
By grooming the traffic on the PSTN side of the PBX all user PBX functions will continue to  
operate. This system will operate in exactly the same way as the phone system worked  
before the gateways were installed, for example, even existing shortcode dialling will  
continue to work (as the PBX will expand this to a full number before presenting it to the  
PSTN).  
- For VoIP users  
VoIP users will be able to dial extension numbers of their ‘home’ PBX as well as PSTN  
numbers. Calls will be routed via the Vega attached to their ‘home’ PBX and directed  
either towards the PBX or the PSTN. As their calls into the PBX appear to the PBX to  
come from the PSTN they will not have advanced functionality that an extension user  
would have (e.g. voice mail, call transfer, conferencing)  
For incoming calls to the home-worker, it is best that the home-worker is given one of the  
DDI numbers available in the ISDN trunk. When the Vega receives a call from the PSTN to  
that phone number it will groom the call off and forward it to the home-user.  
Colleagues on the PBX will have to dial the home-worker’s full number (not just the  
extension number) so that the call is routed by the PBX towards the PSTN.  
Version 2.1  
16 March 2006  
Page 6 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
When the Vega receives a call from the PBX to that phone number it will groom the call off  
and forward it to the home-user.  
USPs  
Vega dial planner capability provides powerful routing decision making. For example, calls  
can be routed based on dialled number prefixes, full numbers, and even on who is making  
the call.  
Considerations  
Check the numbers that are presented to the PSTN consider for example:  
- does the PBX present both National format and local format numbers to the PBX?  
- does the PBX insert any “carrier select prefix” to the dialled number?  
- can “block caller ID” or other prefixes be used in dialled numbers?  
Consider NAT traversal for home worker  
Emergency calls should be routed directly to the PSTN, not over the VoIP link  
Program in fixed number length dial plans where possible to avoid the DTMF timeout  
delays incurred when routing calls to variable length numbers  
Call transfer from PBX to PBX will only work if call transfer PBX to PBX worked across the  
PSTN (the PBX may block this).  
If extension number dialling of the far end PBX or home-workers is to be used (rather than  
just grooming off the full dialled destination number), remember the extension number will  
have to be prefixed with the PBX’s ‘connect to PSTN’ digit; usually 9 in UK, but can be 0 in  
Europe.  
If IP connectivity fails, calls can be programmed to be routed over the PSTN instead; there  
may, however, be a bit of delay in deciding that the IP connection is down.  
Ensure that only authorised home-worker / remote office users can make outbound calls to  
the PSTN – avoid just anyone with SIP connectivity to the internet being able to make calls  
through the Vega and to the PSTN  
Future extensions  
Addition of a Proxy to support additional features for IP users  
If advanced PBX functionality is required, consider the use of analog long line extensions  
Extend the “on net” group to a group of companies who regularly call each other so all  
calls between these companies are free  
Connection to an ITSP for cheaper off-net calls – and possibly for more flexible ingress of  
national, international or non-geographic numbers.  
If PSTN connectivity is not reliable at any site then VoIP can be used as a backup to route  
outbound calls via a different site.  
More than 2 sites to be interconnected  
3rd party product choices  
For NAT traversal between sites, consider SNOM NAT filter, SIParator or Alcatel T610  
Version 2.1  
16 March 2006  
Page 7 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
3. VoIP between sites and home-worker / remote office using a proxy  
Aim: To reduce inter-site call costs without changing the user experience, and adding in more  
advanced home-worker support  
Many multi-site companies find that staff making calls between sites is a significant company  
expense. Finding a method of reducing to zero the ongoing costs of these calls would be a real  
benefit to the bottom line. Also, with more and more people working from home – either  
permanently or just occasionally – it is useful and cost effective to have home-workers able to  
make free calls to office based staff and use the company PSTN connection for outbound  
company calls (no more telephony expenses forms, and appropriate tariffs for outbound calls  
based on those negotiated by the company with their chosen carrier).  
By inserting a Vega gateway between the PSTN and the PBX the Vega can consider how to  
handle every call between the PSTN and the PBX either routing the call over IP, routing the call to  
the PBX or routing the call to the PSTN.  
With the addition of a Proxy / Media server to the system, advanced processing of calls can be  
accomplished. This functionality is Proxy dependent, but often includes, for example  
Forked calls (presenting calls to more than 1 destination at a time) – where a desk phone  
(off the PBX) and the home-office phone (off a residential gateway) both ring when a call is  
received for that extension number  
Road warrior support – where road warriors use SIP devices (residential gateways, or soft  
phones) as their telephone. Wherever the road warrior connects their SIP device it  
registers with the Proxy so that the proxy knows the current IP address of the road warrior.  
Any phone calls can then be sent over IP to this IP address. The road warrior can therefore  
‘be in the office’ wherever they are in the world – at home, in a hotel, or even at a customer  
site.  
Voice-mail for IP devices  
Conferencing server for IP devices  
Follow me, call diversion on no IP endpoint registered  
Other functionality – proxy dependent  
Version 2.1  
16 March 2006  
Page 8 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
This configuration can form a good starting point for migrating to VoIP without immediately losing  
the existing telecoms infrastructure.  
The Vega can be configured to either route specific telephone numbers to the SIP proxy – where  
only a small set of numbers is to be handled by the VoIP system, or all calls can be forwarded to  
the Proxy for it to decide how all calls should be routed.  
Use of a Proxy can also assist in NAT and firewall traversal.  
If a significant proportion of the company’s calls are made between sites then it is possible that  
further cost savings can be made by reducing the number of channels connected to the PSTN.  
Telephony Network  
- For PBXs  
E1 Euro ISDN signalling is supported by Vega 400 and Vega 100  
T1 NI1 & NI2 signalling is supported by Vega 400 and Vega 100  
T1 Loop start, Ground start and Wink start CAS signalling is supported by Vega 400 and  
Vega 100  
BRI Euro ISDN signalling is supported by Vega 50 BRI  
Analog connection is possible, but is not preferred (for details on analog connectivity see  
later section)  
- For home-workers  
Analog FXS gateway to connect to analog phone  
Optional FXO port to allow local connection to PSTN  
IP handset or soft phones may also be used  
IP Network  
Recommend – private IP network between sites  
Can be any reliable IP connection, including leased line, line of site laser / wireless  
connection and satellite  
QOS should be implemented to ensure that VoIP packets pass through the network in a  
consistent and timely manner and are not discarded by congested routers  
Consider VPN between sites as this:  
- removes any issues with firewalls / NAT  
- encrypts audio and management data  
For inter-site operation the Internet can be used, but contention and available bandwidth  
must be considered  
A VPN can be used from the home user to the company LAN. This is important for  
securing data access to the company site, and for VoIP it gets around NAT traversal issues  
User experience  
- For PBX users that are not routed via the Proxy  
By grooming the traffic on the PSTN side of the PBX all user PBX functions will continue to  
operate. This system will operate in exactly the same way as the phone system worked  
before the gateways were installed, for example, even existing shortcode dialling will  
continue to work (as the PBX will expand this to a full number before presenting it to the  
PSTN).  
Version 2.1  
16 March 2006  
Page 9 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
- For home users who are routed via the proxy  
Home-workers will be able to dial extension numbers of any of the PBXs (as the proxy will  
resolve the phone number and route the call to the appropriate Vega gateway). Calls made  
to users on a PBX will appear to the PBX to come from the PSTN and so will not provide  
the advanced functionality to the home-worker that an extension user would have (e.g.  
voice mail, call transfer, conferencing). The advanced features available to the home-  
worker will be those available from the SIP proxy.  
The home-worker will be able to dial PSTN numbers. If the company sites are in different  
areas, and especially if the sites are in different countries the SIP proxy can provide least  
cost routing functionality to route the call to the most appropriate Vega to deliver the call to  
the PSTN.  
For incoming calls to the home-worker, it is best that the home-worker is given one of the  
DDI numbers available in the ISDN trunk. When the Vega receives a call from the PSTN to  
that phone number it will groom the call off and forward it to the home-user.  
Colleagues on the PBX will have to dial the home-worker’s full number (not just the  
extension number) so that the call is routed by the PBX towards the PSTN. When the Vega  
receives a call from the PBX to that phone number it will groom the call off and forward it to  
the home-user.  
- For users who have their calls routed via the proxy  
This enables users to have dual ringing – or multiple parallel ringing, for instance having  
your desk phone (connected to the PBX) and your home-office VoIP phone both ringing  
whenever you receive an incoming call.  
Users can have VoIP pre-processing features of their calls, features like the ability to  
decide which order your phones are called in, and which callers are allowed to disturb you  
and which should be routed through to voice mail.  
- Proxy initiated calls  
Proxy initiated functionality will be available, e.g. where the user can use a web browser to  
access the proxy and drag and drop user icons to initiate, for example a conference  
Calls that were connected using the Proxy will retain the capability to be controlled by the  
proxy, for instance call transfer could be initiated by the user on the web interface to the  
proxy if the original call was routed via the proxy  
USPs  
VegaStream have a wide range of VoIP gateways to provide connections for both home  
and office based workers.  
Considerations  
Check the numbers that are presented to the PSTN consider for example:  
- does the PBX present both National format and local format numbers to the PBX?  
- does the PBX insert any “carrier select prefix” to the dialled number?  
- can “block caller ID” or other prefixes be used in dialled numbers?  
Be careful of interactions between PBX functionality and Proxy functionality, e.g. if Voice-  
mail is enabled on both the PBX and the VoIP system, when does each get the voice-mail,  
and is it acceptable that users may have to check two places to collect their voice-mail?  
Consider NAT traversal for home worker  
The maximum number of parallel ringing calls into the PBX may be limited by the maximum  
number of SIP registrations that the Vega can make.  
Emergency calls should be routed directly to the PSTN, not over the VoIP link  
Version 2.1  
16 March 2006  
Page 10 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
Program in fixed number length dial plans where possible to avoid the DTMF timeout  
delays incurred when routing calls to variable length numbers  
Call transfer from PBX to PBX will only work if call transfer PBX to PBX worked across the  
PSTN (the PBX may block this).  
If extension number dialling of the far end PBX or home-workers is to be used (rather than  
just grooming off the full dialled destination number), remember the extension number will  
have to be prefixed with the PBX’s ‘connect to PSTN’ digit; usually 9 in UK, but can be 0 in  
Europe.  
If IP connectivity fails, calls can be programmed to be routed over the PSTN instead; there  
may, however, be a bit of delay in deciding that the IP connection is down.  
Ensure that only authorised home-worker / remote office users can make outbound calls to  
the PSTN – avoid just anyone with SIP connectivity to the internet being able to make calls  
through the Vega and to the PSTN  
Where a media server is used, consider placing it close to where the media is going to be  
used – this minimises the distance over which ‘fat data pipes’ are needed to route audio  
traffic.  
Future extensions  
Remove PBX and make all users IP users  
Extend the “on net” group to a group of companies who regularly call each other so all  
calls between these companies are free  
Connection to an ITSP for cheaper off-net calls – and possibly for more flexible ingress of  
national, international or non-geographic numbers.  
If PSTN connectivity is not reliable at any site then VoIP can be used as a backup to route  
outbound calls via a different site.  
More than 2 sites to be interconnected  
3rd party product choices  
SNOM, Mitel MKC, IVR technologies, NetSapiens  
Version 2.1  
16 March 2006  
Page 11 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
4. Analog long line extensions  
Aim: To allow 1 or more remote sites / home-workers to have extensions on a centralised PBX,  
retaining all the PBX’s analog phone functionality  
When expanding into new offices it can be expensive and inconvenient to set up a new PBX or  
run analog leased lines from the main site to the new remote site. Data infrastructure between  
sites however is essential. Keeping the Main site PBX for all users brings many benefits, including  
extension number dialling to all personnel, voice mail on a single site, common functionality for all  
phone users (people can move offices and do not have to learn a new phone system).  
By using data connections to join remote users back to the main site PBX, this concept can be  
extended to home users, using the Internet as the data network. Now home-workers have all the  
PBX functionality in their home-offices. Also no more telephone expense claims, business calls  
are all made through the company PBX, and appropriate tariffs for outbound calls based on those  
negotiated by the company with their chosen carrier.  
Telephony Network  
Vega 50 FXO supports up to 10 connections to the central site PBX  
Vega 50 FXS supports up to 8 telephone connections (which can include fax machines and  
modems) plus 2 FXO ports (can be used  
Version 2.1  
16 March 2006  
Page 12 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
IP Network  
Recommend – private IP network between sites  
Can be any reliable IP connection, including leased line, line of site laser / wireless  
connection and satellite  
QOS should be implemented to ensure that VoIP packets pass through the network in a  
consistent and timely manner and are not discarded by congested routers  
Consider VPN between sites as this:  
- removes any issues with firewalls / NAT  
- encrypts audio and management data  
For inter-site operation the Internet can be used, but contention and available bandwidth  
must be considered  
Recommend – a VPN should be used from the home user to the company LAN. This is  
important for securing data access to the company site, and for VoIP it gets around NAT  
traversal issues  
User experience  
Exactly the same operation as if their analog phone was physically connected directly to  
the PBX  
- pick the handset up and you dial tone from the PBX (this may be stuttered by the PBX to  
indicate voice mail)  
- hookflash / recall will allow the phone to access advanced features of the PBX (like call  
transfer, follow me, camp-on-busy, conferencing)  
There might be a slightly longer delay needed between clearing down from 1 call to getting  
dial tone ready to make the next call.  
USPs  
Prolongs the life of existing legacy equipment  
Allows users to be situated where ever most convenient, not limited by the location of the  
PBX.  
PBX features are available to users (through use of hook-flash and DTMF commands)  
No user re-training  
Simple self-contained solution  
Considerations  
Need to ensure that the PBX has analog phone connectivity – proprietary digital phones  
cannot be extended in this manner  
Emergency calls must be considered; it may be necessary to have a special “Emergency”  
phone  
Future extensions  
Installation of Proxy / gatekeeper to provide IP-PBX functionality (FXS gateways and IP  
phones can be re-used in this new configuration).  
3rd party product choices  
For NAT traversal between sites, consider SNOM NAT filter, SIParator or Alcatel T610  
For VPN at home site consider Alcatel T610  
Version 2.1  
16 March 2006  
Page 13 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
5. IP Telephony  
Aim: To gain additional features available through use of VoIP technology  
VoIP offers users many facilities and many features currently not known with PBX telephony  
solutions. Much of the functionality offered by VoIP is controlled through access to the proxy /  
gatekeeper / media server using a web browser on a PC. Both IP phones and analog phones (via  
an analog VoIP gateway) may be used to provide telephone access to the users. Intra company  
calling is fully VoIP, as are calls to and from VoIP enabled home-workers. Calls to and from the  
PSTN need conversion to/from traditional protocols, and this is achieved using Vega trunking  
gateways.  
Many ACD solutions are also moving to VoIP infrastructures. The use of VoIP makes it easy to  
distribute agents across multiple sites and out to home-workers. By addressing calls and screen  
pops to IP addresses it is up to the IP data infrastructure to deliver the messages to the correct  
destinations whether it be local or distant.  
Telephony Network  
- For connection to PSTN  
E1 Euro ISDN signalling is supported by Vega 400 and Vega 100  
T1 NI1 & NI2 signalling is supported by Vega 400 and Vega 100  
T1 Loop start, Ground start and Wink start CAS signalling is supported by Vega 400 and  
Vega 100  
BRI Euro ISDN signalling is supported by Vega 50 BRI  
Analog connection is possible, but is not preferred (for details on analog connectivity see  
later section)  
- For end users  
Analog FXS gateways allow connection to analog phone  
IP handsets or soft phones may also be used  
Version 2.1  
16 March 2006  
Page 14 of 33  
Vega Gateway Scenarios.  
©2005-2006 VegaStream Ltd.  
Examples are given in best faith – ensure that you check the capabilities of systems before  
deploying them, especially the functionality of devices not designed and delivered by VegaStream.  
Download from Www.Somanuals.com. All Manuals Search And Download.  
IP Network  
Recommend – private IP network between sites  
Can be any reliable IP connection, including leased line, line of site laser / wireless  
connection and satellite  
QOS should be implemented to ensure that VoIP packets pass through the network in a  
consistent and timely manner and are not discarded by congested routers  
Consider VPN between sites as this:  
- removes any issues with firewalls / NAT  
- encrypts audio and management data  
A VPN can be used from the home user to the company LAN. This is important for  
securing data access to the company site, and for VoIP it gets around NAT traversal issues  
User experience  
The user experience will be completely controlled by the SIP proxy / H.323 gatekeeper.  
Choice of the correct application server device will be key to successful delivery of service.  
USPs  
Vega gateways are very good at interoperating with 3rd party VoIP devices.  
Features available that are not available on traditional telephony  
Considerations  
Check that the data network is capable of supporting VoIP (QoS / bandwidth)  
Where a media server is used, consider placing it close to where the media is going to be  
used – this minimises the distance over which ‘fat data pipes’ are needed to route audio  
traffic.  
Future extensions  
Connection to an ITSP for cheaper off-net calls – and possibly for more flexible ingress of  
national, international or non-geographic numbers.  
3rd party product choices  
PBX: Snom  
Intelligent Call Routing: Rostrvm, Teleware, Alceo, Interactive intelligence  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
6. PSTN toll bypass  
Aim: To use internet to provide cheap inter-country calling  
Vega gateways, together with Application servers supporting pre-paid card services allow  
Telephony Network  
E1 Euro ISDN signalling is supported by Vega 400 and Vega 100  
T1 NI1 & NI2 signalling is supported by Vega 400 and Vega 100  
T1 Loop start, Ground start and Wink start CAS signalling is supported by Vega 400 and  
Vega 100  
BRI Euro ISDN signalling is supported by Vega 50 BRI  
Analog connection is possible, but is not preferred (for details on analog connectivity see  
later section)  
IP Network  
Recommend – private IP network between sites  
Can be any reliable IP connection, including leased line, satellite and the internet  
QOS should be implemented if possible to ensure that VoIP packets pass through the  
network in a consistent and timely manner and are not discarded by congested routers  
The Internet is often used for carrying the VoIP traffic; contention and available bandwidth  
must be considered  
User experience  
The caller dials a local, often toll-free or lo-call telephone number and gets routed to a VoIP  
gateway. The call is answered by the Application Server and the user is asked to enter  
account number, pin and destination phone number. If the caller has sufficient credit the  
call is routed through to the most appropriate (Least Cost Routed) destination VoIP  
gateway and the gateway presents the call over the PSTN to the required destination  
number. If credit runs out during a call then the call can be torn down (terminated).  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
USPs  
Vega gateways have been shown to have fast call set-up times and increased call hold  
times compared to other manufacturers’ gateways due to the quality of the connected call.  
Considerations  
Use of analog gateways to break into / break out of the PSTN may have timing issues over  
start and end of calls unless line current reversal analog signalling is used.  
Future extensions  
Expansion to a wider range of destination countries and cities through use of ITSP  
connectivity.  
3rd party product choices  
IVR technologies  
Tangerine Inc  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
7. TDM to / from VoIP converter  
Aim: To allow existing manufacturers of TDM based equipment (switches / IVR systems etc) to  
have a VoIP offering.  
Many manufacturers of TDM equipment are being asked by customers about their policy on VoIP.  
Other customers are more demanding and informing suppliers that a migration path to VoIP is  
essential. It is expensive to integrate a full VoIP solution into a product ‘just because a customer  
has asked’. Use of Vega gateways allows integration of VoIP to existing TDM equipment on an as  
needed basis.  
The Vega gateway connects to the E1, T1 or BRI interface of the existing equipment and can  
receive calls and convert them to SIP or H.323; the Vega gateway can also receive SIP or H.323  
calls and convert them to E1, T1 or BRI calls.  
Telephony Network  
E1 Euro ISDN signalling and QSIG signalling is supported by Vega 400 and Vega 100  
T1 NI1 & NI2 signalling is supported by Vega 400 and Vega 100  
T1 Loop start, Ground start and Wink start CAS signalling is supported by Vega 400 and  
Vega 100  
BRI Euro ISDN signalling is supported by Vega 50 BRI  
Analog connection is also possible (for details on analog connectivity see later section)  
IP Network  
Use of SIP or H.323 is defined by the code loaded into the Vega gateway  
User experience  
The user will now be able to use the functionality that they had achieved using TDM  
connectivity over a VoIP interface  
USPs  
Vega gateways are designed to support full system loading, so performance will not  
degrade as the unit becomes fully loaded  
A Vega gateway solution is cheaper than a PC card based solution, and uses dedicated  
hardware rather than general-purpose hardware.  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
Considerations  
Not all supplementary services supported in ISDN signalling directly map to VoIP  
protocols. If supplementary service are needed it is important to check that they will map  
OK.  
Future extensions  
-  
3rd party product choices  
-  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
8. Leased line eliminator for QSIG connections between sites  
Aim: To save leased line call costs between sites  
In larger companies, the PBXs on multiple sites may be linked together via leased lines and using  
QSIG signalling in order to make the distributed PBXs function as though they were a single PBX.  
When linking PBXs together in this way, point-to-point connections are required between each  
and every PBX. This becomes expensive in terms of leased lines.  
By using the existing company IP network and Vega gateways that support ‘tunnelled’ QSIG, the  
leased lines are no longer required. The Vega gateways receive the QSIG information on their T1  
or E1 interfaces – then, using the destination information in the QSIG header the Vega can select  
the correct far end Vega gateway to send the ‘tunnelled’ QSIG data to. The far end gateway  
unpacks the received ‘tunnelled’ QSIG messages and presents them on its T1 or E1 interface.  
The PBXs do not know that the QSIG connection was not a leased line.  
Telephony Network  
E1 ‘tunnelled’ QSIG signalling is supported by Vega 400 and Vega 100  
T1 ‘tunnelled’ QSIG signalling is supported by Vega 400 and Vega 100  
IP Network  
Recommend – private IP network between sites  
Can be any reliable IP connection, including leased line and line of site laser / wireless  
connection  
Satellite IP connectivity is a possibility, though a system like this should be tested to ensure  
that QSIG timeouts are not exceeded by the satellite latency.  
QOS should be implemented to ensure that VoIP packets pass through the network in a  
consistent and timely manner and are not discarded by congested routers  
For inter-site operation the Internet can be used, but contention and available bandwidth  
must be considered  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
Consider VPN between sites as this:  
- removes any issues with firewalls / NAT  
- encrypts audio and management data  
Vega gateways use the ECMA 333 standard for tunnelling QSIG  
User experience  
There is no change to the existing user experience. The information provided between  
PBXs is identical, so the PBXs see no change at all.  
The user experience is defined only by the PBX capabilities.  
USPs  
QSIG signalling and QSIG-like signalling are supported by the Vega  
Considerations  
Point-to-point trunking1 rather than fully meshed2 connection of Vega gateways may be  
needed if:  
- the protocol is QSIG-like and not fully QSIG compatible (some manufacturers use QSIG  
as a basis for their inter-PBX connectivity, but retain certain proprietary elements)  
- PBXs send calls for specific destination PBXs to specific trunks  
QSIG from different manufacturers may have proprietary elements. If QSIG PBXs would  
not talk together when connected with a leased line, they will not operate together when  
connected using VegaStream QSIG tunnelling.  
Future extensions  
-  
3rd party product choices  
-  
1 ‘point-to-point trunking’ means that all calls on a specific Vega trunk will be routed to a specific destination trunk on a specific  
destination Vega gateway  
2 ‘fully meshed’ indicates that on a call by call basis the Vega gateway can select the appropriate IP endpoint to send the QSIG  
data to.  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
9. Analog connections  
Where PBXs or incoming lines are analog rather than PRI or BRI, Voice over IP can still be used.  
Care must be taken however, because unlike the symmetrical operation of digital interfaces –  
where any information provided by one digital telephony device to the other digital telephony  
device can also be passed in the opposite direction, analog signalling is not symmetric. The two  
different ends of an analog line are called FXS and FXO. There are therefore two different types of  
analog VoIP gateway interface, FXS and FXO. The choice of FXS or FXO interface will depend on  
both what the gateway needs to connect to and also the functionality that it needs to support.  
FXS: An analog PSTN line and an analog extension interface to a PBX are both FXS interfaces –  
analog telephones may be directly into FXS interfaces.  
It supplies a DC voltage to the line (around 48v).  
It alerts the attached device that there is a call available by supplying ‘ringing voltage’ on  
the line. It does not provide any information about the number that was dialled to make  
this line ring.  
Caller ID may be provided3; this is typically provided by an FSK (modem) tone burst  
between first and second rings.  
Typically it cannot indicate that the call has cleared, though there are extensions to the  
analog signalling specification (loop current disconnect / battery stop & line current reversal  
/ battery reversal) that do allow a physical indication of call cleardown.  
The FXS interface detects that a new call is being sent to it by detecting that a current,  
‘line-current’ is flowing. This line current is triggered by the far end device going ‘off-hook’.  
FXS devices detect DTMF (Dual Tone Multi Frequency) tones which are used for dialling  
telephone numbers.  
FXS devices typically detect hook-flash (also known as recall); hook-flash is used by PBXs  
and other telephony systems to alert them and trigger them to move into a command mode  
(e.g. to initiate call transfer or conferencing when in the middle of a call).  
FXS devices detect the end of the call by loss of ‘line-current’ when the far end device  
goes ‘on-hook’.  
FXO: An analog telephone (which can plug into a PSTN line or plug into an analog extension  
interface of a PBX) is an FXO device.  
It may use the line voltage supplied by the FXS line to power low current circuitry  
It detects that an incoming call is being presented to it by receiving ‘ringing voltage’ – it  
does not receive any indication of the number that was dialled for the call to reach here.  
Caller ID may be received4; this is typically provided by an FSK (modem) tone burst  
between first and second rings.  
If the FXS device supports loop current disconnect / battery stop or line current reversal /  
battery reversal signalling, the Vega 50 FXO can be configured to detect call cleardown at  
the end of the call – otherwise the call must be cleared from the VoIP side. (N.B. this  
means that FXO to FXO calls – even across a VoIP link are not appropriate if neither  
cleardown method is supported at either end).  
To initiate a call the FXO device goes ‘off-hook’ – actually it completes a circuit allowing  
current to flow between the FXS and FXO.  
To dial a telephone number the FXO device uses DTMF tones.  
3 Vega 50 FXS supports Belcore sdmf and mdf standards, SPA-1001, SPA-2000, SPA-3000 support these and many others  
4 Vega 50 FXO supports Belcore sdmf and mdf standards, SPA-2000 support these and many others  
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Hook-flash (also known as recall) can be generated by Vega 50 FXO (not SPA-2000) in  
order to request the attached FXS to switch to command mode (e.g. to initiate a call  
transfer on a PBX extension port)  
To clear a call the FXO goes ‘on-hook’ – actually it breaks the circuit that was made when  
the device went ‘off-hook’  
The key thing to note is: FXS devices can receive dialled number information but cannot provide it  
and FXO devices can generate dialled number information but cannot receive it.  
Analog applications:  
9.a Vega FXS for analog telephones  
Analog telephones can be connected directly into the Vega FXS ports. This allows existing or  
new (and often cheap) analog telephones to be connected into a VoIP system. This can be  
especially useful where Cat 5 cabling already runs analog phones to peoples’ desks; the Vega can  
replace the PBX in the telecoms cabinet, the Cat 5 telephone extensions terminate on the Vega  
FXS rather than on the old PBX.  
A major benefit to users is that they see no change to their working environment and equipment.  
Also in industrial environments, ruggedised analog phones or cheap disposable phones are much  
more appropriate user devices than the more expensive, non-ruggedised IP phones.  
Vega 50 FXS (SIP) can initiate call transfer (blind and consultative), put a call on hold, and can  
toggle between a caller on hold and a second called party through use of hookflash and DTMF  
keys on the attached telephones.  
Considerations  
Message waiting indicator lamp operation is not supported – message waiting indication  
using stuttered dial tone can be used.  
Caller ID is only supplied in sdmf or mdmf formats on the Vega 50 FXS (our range of home  
gateways support more formats).  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
9.b Analog breakout to the PSTN – e.g. for a company or for a PSTN toll bypass  
A Vega FXO can be connected directly to PSTN lines.  
Calls from VoIP into the PSTN specify the desired destination number by sending DTMF tones  
into the PSTN.  
Considerations  
Disconnect supervision is only supported by physical signalling – line current reversal or  
loop current disconnect; no voice or tone detect can be configured on the Vega to indicate  
end of call  
Answer supervision is only supported by physical signalling – line current reversal  
Calls received on the telephony interface arrive with no dialled number information as  
analog signalling just uses a ringing voltage to indicate the call arrival.  
Different impedances are required for ‘approved’ connection in different countries. Europe  
should have the Vega FXO interface configured for CTR21 impedance; the US should have  
the interface configured for 600R operation. 900R impedance is also available.  
Version 2.1  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
9.c FXS connection to a PBX  
Considerations  
Out-dialled calls from the PBX can pass a dialled number to the Vega; it can use this to  
decide which destination VoIP gateway to route the call to (i.e. multiple destination  
gateways may be supported), and also which destination end-point to route the call to.  
Calls from the Vega to the PBX, are like calls from the PSTN – no dialled number is passed,  
so typically the call will be routed to a receptionist or an auto attendant (or the FXS  
gateway may just be used for outbound calls)  
May not be able to receive transferred calls – some PBXs will prevent calls being  
transferred to a trunk interface.  
9.d) FXO connection to a PBX  
When making calls from VoIP to PBX the dialled number is passed using DTMF tones played into  
the PBX.  
Considerations  
An FXO device on an extension interface of a PBX can accept transferred calls, but as no  
dialled number is presented to the gateway by the PBX, the gateway must be configured  
with a static destination to route the call to.  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
9.e PBX to PBX connectivity – using both FXS and FXO  
In this configuration calls may be routed PBX to FXS to FXO to PBX or  
PBX to FXO to FXS to PBX.  
Calls FXS to FXO  
Outdialled calls from the PBX can pass a dialled number to the Vega; it can use this to decide  
which destination VoIP gateway to route the call to (i.e. multiple destination gateways may be  
supported), and also which destination end-number to route the call to.  
Considerations  
Call transfer from a PBX extension to the other PBX may not be blocked by the PBX if the  
local PBX does not allow call transfers to Trunk Interfaces  
Calls FXO to FXS  
To connect to the far end PBX, call the local PBX extension number of a line connected to the  
FXO gateway. The FXO gateway will route the call to the destination FXS gateway.  
Considerations  
Calls presented PBX to FXO can only provide ringing voltage to indicate call arrival. The  
destination VoIP gateway to deliver the call to is defined statically in the Vega FXO.  
Calls presented FXS to PBX can only provide ringing voltage to indicate call arrival. Just  
as PSTN calls to analog trunk interfaces of a PBX have to be routed to an operator or auto  
attendant, so do calls from the Vega FXS.  
Enable disconnect supervision on FXO / PBX interface if possible.  
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9.f PBX to PBX connectivity – using FXS only  
Out-dialled calls from the PBX can pass a dialled number to the Vega; it can use this to decide  
which destination VoIP gateway to route the call to (i.e. multiple destination gateways may be  
supported).  
Considerations  
Call transfer from a PBX extension to the other PBX may not be blocked by the PBX if the  
local PBX does not allow call transfers to Trunk Interfaces  
Calls presented FXS to PBX can only provide ringing voltage to indicate call arrival. Just  
as PSTN calls to analog trunk interfaces of a PBX have to be routed to an operator or auto  
attendant, so do calls from the Vega FXS.  
9.g PBX to PBX connectivity – using FXO only  
To connect to the far end PBX, call the local PBX extension number of a line connected to the  
FXO gateway. Calling this local extension number will trigger the local FXO gateway to  
immediately set up a VoIP call to the far end FXO gateway. The caller will receive dial tone from  
the far end PBX – the caller can then dial the desired destination number into the far end PBX.  
Considerations  
Need disconnect supervision on at least 1 Vega – preferably both.  
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9.h Mass analog connection – using a Vega 400 and a channel bank  
Where mass capacity of FXS and / or FXO interfaces are required it can be cheaper to use a Vega  
400 together with a channel bank.  
Considerations  
Ensure that the channel bank supports the required functionality, e.g. will it support hook-  
flash, call cleardown detection.  
9.i Analog long line extensions  
See “1 VoIP between sites” above.  
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10. Appendix 1 – 3rd Party Products  
This list of 3rd party products is by no means a complete list of devices that Vega gateways have  
been tested with and interoperate with, but it provides an indication of the range of products  
available.  
The products are listed in alphabetical order:  
Alcatel Thomson SpeedTouch 610 router  
+ built in VPN connectivity  
+ optional SIP proxy  
- only basic proxy functionality, no “intelligence” features  
Alceo  
ArelNet Softswitch  
Asterisk SIP proxy  
+ freeware - free  
+ freeware – proxy is continually being developed  
- freeware – if you use it you have to support it; it does not come with a manufacturer’s backing  
+ as a developer / VoIP reseller it can prove to be a good base architectire on which to add your  
own tailored features.  
- when used with audio cards it runs out of processing power quickly  
Axiom provisioning system  
Broadsoft  
+ long-standing large, highly reliable SIP proxy  
- expensive  
Brekeke OnDO  
+ low cost of entry – approx $300 for trial  
+ low cost purchase – approx $3,000  
+ IP PBX and SIP proxy variants  
Centile  
Cirpack  
DynamicSoft  
FrontRange IPCC  
+ ACD call queuing functionality  
+ built in media server  
- only designed for use with Heat / Goldmine  
- cost per seat  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
Genesys  
+ ACD with integrated CRM solution  
GNU H.323 GateKeeper  
Ingate SIParator  
+ Standalone unit  
+ local NAT traversal (works with or in parallel with existing firewall)  
+ option to use as firewall as well as SIP proxy  
+ far end NAT traversal  
- Does not support far end NAT traversal with Sipura code  
- only basic proxy functionality, no “intelligence” features  
Interactive Intelligence  
+ ACD call queuing functionality  
- ? cost  
IVR Technologies – Talking SIP  
LignUp SIP Proxy  
Mitel Knowledge MKC proxy  
NetCentrex Gatekeeper  
NetSapiens V-Box  
+ Standalone unit, no hard disc  
Nortel CS1000  
Nortel MCS5100  
Nortel MCS5200  
Nuera SSC  
Polycom phones  
+ good quality  
+ good value  
PBXnSIP  
+ Well functioned IPBX  
- needs to be run on a PC  
Rostrvm  
+ ACD call queuing functionality  
+ operates on CTI and SIP so can manage calls in the PBX and VoIP calls  
- ? cost  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
Siemens’ OpenScape  
SIPExchange (PingTel)  
+ SIP aware firewall  
SonicWall  
+ SIP aware firewall  
+ built in VPN tunnelling functionality (multiple tunnels)  
+ can receive SIP INVITES and forward them to the appropriate internal VoIP devices  
+ easy to configure  
SNOM 4S, proxy, media server, Nat traversal filter  
+ Proxy, media server and NAT traversal available  
- needs to be run on a / some windows / linux PCs  
+ Standalone version available  
Tangerine Inc SIP / H.323 application server  
Teleware  
+ ACD functionality  
Session Border controllers  
For larger installations and ISP installations there are devices that sit in the Internet called session  
border controllers which (amongst other things) resolve NAT traversal issues. These are typically  
high capacity and expensive. They are manufactured by, for example, Jasomi, Kagoor, and  
Nextone.  
Also Sansay  
Other Devices  
Expand Networks have the Expand IP Accelerator which allows IP data from many applications to  
be compressed across a point to point link, allowing more bandwidth for VoIP data. They can  
also provide QoS marking of data to prioritise various types of information.  
Bandwidth management routers allow bandwidth to be reserved for VoIP.  
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Download from Www.Somanuals.com. All Manuals Search And Download.  
ITSPs and VoIP services  
Call UK  
Engin  
GossipTel  
Gradwell  
Hipcom  
Primus  
Pulver free world dial up  
Telic.net  
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11.Appendix 2 – Things to consider  
Audio quality  
Ensure that the IP network does not drop data  
Use QOS to ensure that VoIP traffic is given priority through routers  
Choose best quality codec that is possible for the bandwidth available  
Turn off silence suppression  
Turn off Echo cancellation  
Existing firewalls  
Ensure that the existing firewall / NAT can cope with the data bandwidth required for the  
voice traffic (ensure that they do not drop packets, introduce too much latency or introduce  
too much jitter). If it cannot, consider using a specialist VoIP firewall for handling just the  
VoIP traffic.  
Beware, some VoIP aware firewalls ‘get it wrong’. A VoIP-aware firewall that mis-translates  
things can be worse than a non VoIP-aware firewall. Check that the VoIP-aware firewall  
works correctly or turn off the functionality and use a purpose-designed product.  
Download from Www.Somanuals.com. All Manuals Search And Download.  

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