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Multi Site
- Thread starter monacom
- Start date
D) Two sites connected via a LAN Link
Lets assume we want to connect two sites together via a LAN link and use VoIP to place calls between sites. This configuration only allows calls to be placed between sites (i.e. internal calls only).
The ArgentBranch at Site A is on network 192.168.131.0. The IP address of the unit is 192.168.131.1. All extensions start with a 7 e.g. 701, 702.
The ArgentBranch at Site B is on network 192.168.43.0. The IP address of the unit is 192.168.43.1. All extensions start with an 8 e.g. 801, 802.
Site A
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.43.1 (IP address of Branch at Site B)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.43.0 (network address of Site B)
IP Mask = 255.255.255.0 (subnet mask of Site B)
Destination = LAN2
3 Create a short code to route all calls starting with an 8 to Site B via the VPN line created above :-
Shortcode = 8N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Site B
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.131.1 (IP address of Branch at Site A)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.131.0 (network address of Site A)
IP Mask = 255.255.255.0 (subnet mask of Site A)
Destination = LAN2
3 Create a short code to route all calls starting with an 7 to Site A via the VPN line created above:-
Shortcode = 7N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Calling Site A from Site B: -
All you need to do is dial the extension number of the person you require at Site A (e.g. 706). As soon as you dial a 7 the call will be routed via the VPN line via the 7N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 706).
Calling Site B from Site A: -
All you need to do is dial the extension number of the person you require at Site B (e.g. 811). As soon as you dial an 8 the call will be routed the VPN line via the 8N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 811).
Voice over IP - Hints & Tips
The Voice over IP (VoIP) functionality is only available on units supporting this facility and that are running v2 operational software. The maximum number of VoIP channels supported by the unit can be compressed using the optional Voice Compression Module (VCM). This module can support 5 channel, 10 channel or 20 channel. The VCM can be used to compress voice down to either 6k3 (G723)or 8k (G729/Netcoder) and provide echo cancellation (required for high latency circuits).
All channels to a remote destination (IP address) must use the same characteristics i.e. same speed and same mode of compression.
The bandwidth required for a VoIP call is made up of two parts, one of which is the due to the actual digitisation of the analogue voice the other is required by the protocol which is used to wrap the digitised voice up in and transport it to the remote site. VoIP calls require an overhead of 28 bytes per packet (UDP/IP Header overhead) this overhead is increased on a LAN by a further 12 bytes (Ethernet Header) or by 7 bytes over a PPP WAN link.
When transporting voice over low speed links (WANs) it is possible that normal data packets (e.g. 1500 byte IP packets) can prevent or delay the voice data from getting across the link. This can cause a very unacceptable speech quality. Thus it is vital that the routers in the network that carry voice have some form of Quality of Service mechanism (QoS). Version 2 supports the DiffServ (RFC 2474) Quality of Service mechanisms which are based upon the Type of Service (ToS) field in the IP header. The software will prioritise voice, fragment large packets and provide VoIP header compression to minimise the WAN overhead. Typically the VoIP WAN overhead is 35 bytes on 20 byte payload this is 175% overhead! On the WAN protocol this is reduced to only 6 bytes (3 bytes data , 2 bytes CRC and 1 byte HDLC flag) for the same 20 byte packet. The overhead is thus only 30% - a saving of 145%. When calculating the actual link speeds required to support voice traffic you must remember to include the aforementioned overhead. For example an 8Kbps compression voice channel actually requires 10.4Kbps of WAN bandwidth when using QoS or 22Kbps if using standard non QoS routers. QoS routers are also required to ensure low speech latency and to maintain sufficient audible quality. Therefore care should be taken when using third party routers to connect the two networks together.
From the above information it can therefore be deduced that VoIP functionality can be provided by the following:
If VCMs are fitted on the units at both ends then the voice can be encoded at 8k(G729), 6k3(G723), 8k(Netcoder) and 64k(A or u law), otherwise the voice will be encoded at 64k(A or u law). In the first instance the link between the two units can be 64k bps, or greater. The latter requires a link that is greater than 64k bps.
D) Two sites connected via a LAN Link
Lets assume we want to connect two sites together via a LAN link and use VoIP to place calls between sites. This configuration only allows calls to be placed between sites (i.e. internal calls only).
The ArgentBranch at Site A is on network 192.168.131.0. The IP address of the unit is 192.168.131.1. All extensions start with a 7 e.g. 701, 702.
The ArgentBranch at Site B is on network 192.168.43.0. The IP address of the unit is 192.168.43.1. All extensions start with an 8 e.g. 801, 802.
Site A
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.43.1 (IP address of Branch at Site B)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.43.0 (network address of Site B)
IP Mask = 255.255.255.0 (subnet mask of Site B)
Destination = LAN2
3 Create a short code to route all calls starting with an 8 to Site B via the VPN line created above :-
Shortcode = 8N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Site B
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.131.1 (IP address of Branch at Site A)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.131.0 (network address of Site A)
IP Mask = 255.255.255.0 (subnet mask of Site A)
Destination = LAN2
3 Create a short code to route all calls starting with an 7 to Site A via the VPN line created above:-
Shortcode = 7N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Calling Site A from Site B: -
All you need to do is dial the extension number of the person you require at Site A (e.g. 706). As soon as you dial a 7 the call will be routed via the VPN line via the 7N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 706).
Calling Site B from Site A: -
All you need to do is dial the extension number of the person you require at Site B (e.g. 811). As soon as you dial an 8 the call will be routed the VPN line via the 8N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 811).
Voice over IP - Hints & Tips
The Voice over IP (VoIP) functionality is only available on units supporting this facility and that are running v2 operational software. The maximum number of VoIP channels supported by the unit can be compressed using the optional Voice Compression Module (VCM). This module can support 5 channel, 10 channel or 20 channel. The VCM can be used to compress voice down to either 6k3 (G723)or 8k (G729/Netcoder) and provide echo cancellation (required for high latency circuits).
All channels to a remote destination (IP address) must use the same characteristics i.e. same speed and same mode of compression.
The bandwidth required for a VoIP call is made up of two parts, one of which is the due to the actual digitisation of the analogue voice the other is required by the protocol which is used to wrap the digitised voice up in and transport it to the remote site. VoIP calls require an overhead of 28 bytes per packet (UDP/IP Header overhead) this overhead is increased on a LAN by a further 12 bytes (Ethernet Header) or by 7 bytes over a PPP WAN link.
When transporting voice over low speed links (WANs) it is possible that normal data packets (e.g. 1500 byte IP packets) can prevent or delay the voice data from getting across the link. This can cause a very unacceptable speech quality. Thus it is vital that the routers in the network that carry voice have some form of Quality of Service mechanism (QoS). Version 2 supports the DiffServ (RFC 2474) Quality of Service mechanisms which are based upon the Type of Service (ToS) field in the IP header. The software will prioritise voice, fragment large packets and provide VoIP header compression to minimise the WAN overhead. Typically the VoIP WAN overhead is 35 bytes on 20 byte payload this is 175% overhead! On the WAN protocol this is reduced to only 6 bytes (3 bytes data , 2 bytes CRC and 1 byte HDLC flag) for the same 20 byte packet. The overhead is thus only 30% - a saving of 145%. When calculating the actual link speeds required to support voice traffic you must remember to include the aforementioned overhead. For example an 8Kbps compression voice channel actually requires 10.4Kbps of WAN bandwidth when using QoS or 22Kbps if using standard non QoS routers. QoS routers are also required to ensure low speech latency and to maintain sufficient audible quality. Therefore care should be taken when using third party routers to connect the two networks together.
From the above information it can therefore be deduced that VoIP functionality can be provided by the following:
If VCMs are fitted on the units at both ends then the voice can be encoded at 8k(G729), 6k3(G723), 8k(Netcoder) and 64k(A or u law), otherwise the voice will be encoded at 64k(A or u law). In the first instance the link between the two units can be 64k bps, or greater. The latter requires a link that is greater than 64k bps.
![[lightsaber]](/data/assets/smilies/lightsaber.gif "[lightsaber] [lightsaber]")
Lets assume we want to connect two sites together via a LAN link and use VoIP to place calls between sites. This configuration only allows calls to be placed between sites (i.e. internal calls only).
The ArgentBranch at Site A is on network 192.168.131.0. The IP address of the unit is 192.168.131.1. All extensions start with a 7 e.g. 701, 702.
The ArgentBranch at Site B is on network 192.168.43.0. The IP address of the unit is 192.168.43.1. All extensions start with an 8 e.g. 801, 802.
Site A
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.43.1 (IP address of Branch at Site B)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.43.0 (network address of Site B)
IP Mask = 255.255.255.0 (subnet mask of Site B)
Destination = LAN2
3 Create a short code to route all calls starting with an 8 to Site B via the VPN line created above :-
Shortcode = 8N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Site B
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.131.1 (IP address of Branch at Site A)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.131.0 (network address of Site A)
IP Mask = 255.255.255.0 (subnet mask of Site A)
Destination = LAN2
3 Create a short code to route all calls starting with an 7 to Site A via the VPN line created above:-
Shortcode = 7N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Calling Site A from Site B: -
All you need to do is dial the extension number of the person you require at Site A (e.g. 706). As soon as you dial a 7 the call will be routed via the VPN line via the 7N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 706).
Calling Site B from Site A: -
All you need to do is dial the extension number of the person you require at Site B (e.g. 811). As soon as you dial an 8 the call will be routed the VPN line via the 8N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 811).
Voice over IP - Hints & Tips
The Voice over IP (VoIP) functionality is only available on units supporting this facility and that are running v2 operational software. The maximum number of VoIP channels supported by the unit can be compressed using the optional Voice Compression Module (VCM). This module can support 5 channel, 10 channel or 20 channel. The VCM can be used to compress voice down to either 6k3 (G723)or 8k (G729/Netcoder) and provide echo cancellation (required for high latency circuits).
All channels to a remote destination (IP address) must use the same characteristics i.e. same speed and same mode of compression.
The bandwidth required for a VoIP call is made up of two parts, one of which is the due to the actual digitisation of the analogue voice the other is required by the protocol which is used to wrap the digitised voice up in and transport it to the remote site. VoIP calls require an overhead of 28 bytes per packet (UDP/IP Header overhead) this overhead is increased on a LAN by a further 12 bytes (Ethernet Header) or by 7 bytes over a PPP WAN link.
When transporting voice over low speed links (WANs) it is possible that normal data packets (e.g. 1500 byte IP packets) can prevent or delay the voice data from getting across the link. This can cause a very unacceptable speech quality. Thus it is vital that the routers in the network that carry voice have some form of Quality of Service mechanism (QoS). Version 2 supports the DiffServ (RFC 2474) Quality of Service mechanisms which are based upon the Type of Service (ToS) field in the IP header. The software will prioritise voice, fragment large packets and provide VoIP header compression to minimise the WAN overhead. Typically the VoIP WAN overhead is 35 bytes on 20 byte payload this is 175% overhead! On the WAN protocol this is reduced to only 6 bytes (3 bytes data , 2 bytes CRC and 1 byte HDLC flag) for the same 20 byte packet. The overhead is thus only 30% - a saving of 145%. When calculating the actual link speeds required to support voice traffic you must remember to include the aforementioned overhead. For example an 8Kbps compression voice channel actually requires 10.4Kbps of WAN bandwidth when using QoS or 22Kbps if using standard non QoS routers. QoS routers are also required to ensure low speech latency and to maintain sufficient audible quality. Therefore care should be taken when using third party routers to connect the two networks together.
From the above information it can therefore be deduced that VoIP functionality can be provided by the following:
If VCMs are fitted on the units at both ends then the voice can be encoded at 8k(G729), 6k3(G723), 8k(Netcoder) and 64k(A or u law), otherwise the voice will be encoded at 64k(A or u law). In the first instance the link between the two units can be 64k bps, or greater. The latter requires a link that is greater than 64k bps.
D) Two sites connected via a LAN Link
Lets assume we want to connect two sites together via a LAN link and use VoIP to place calls between sites. This configuration only allows calls to be placed between sites (i.e. internal calls only).
The ArgentBranch at Site A is on network 192.168.131.0. The IP address of the unit is 192.168.131.1. All extensions start with a 7 e.g. 701, 702.
The ArgentBranch at Site B is on network 192.168.43.0. The IP address of the unit is 192.168.43.1. All extensions start with an 8 e.g. 801, 802.
Site A
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.43.1 (IP address of Branch at Site B)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.43.0 (network address of Site B)
IP Mask = 255.255.255.0 (subnet mask of Site B)
Destination = LAN2
3 Create a short code to route all calls starting with an 8 to Site B via the VPN line created above :-
Shortcode = 8N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Site B
1 Configure the line that is to be used by the VoIP calls:-
Create a new Line:-
Line Number = 00 (must be different from existing lines)
Line Group ID = 1 (must be different from existing lines)
IP Address = 192.168.131.1 (IP address of Branch at Site A)
Compression Mode = Transparent 64K
2 Create an IP Route:-
IP Address = 192.168.131.0 (network address of Site A)
IP Mask = 255.255.255.0 (subnet mask of Site A)
Destination = LAN2
3 Create a short code to route all calls starting with an 7 to Site A via the VPN line created above:-
Shortcode = 7N
Telephone Number = .
Line Group ID = 1
Feature = Dial
Calling Site A from Site B: -
All you need to do is dial the extension number of the person you require at Site A (e.g. 706). As soon as you dial a 7 the call will be routed via the VPN line via the 7N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 706).
Calling Site B from Site A: -
All you need to do is dial the extension number of the person you require at Site B (e.g. 811). As soon as you dial an 8 the call will be routed the VPN line via the 8N short code configured at Site B. All the numbers dialled will then be passed onto the remote PBX. The remote PBX will then route the call to the appropriate extension (i.e. 811).
Voice over IP - Hints & Tips
The Voice over IP (VoIP) functionality is only available on units supporting this facility and that are running v2 operational software. The maximum number of VoIP channels supported by the unit can be compressed using the optional Voice Compression Module (VCM). This module can support 5 channel, 10 channel or 20 channel. The VCM can be used to compress voice down to either 6k3 (G723)or 8k (G729/Netcoder) and provide echo cancellation (required for high latency circuits).
All channels to a remote destination (IP address) must use the same characteristics i.e. same speed and same mode of compression.
The bandwidth required for a VoIP call is made up of two parts, one of which is the due to the actual digitisation of the analogue voice the other is required by the protocol which is used to wrap the digitised voice up in and transport it to the remote site. VoIP calls require an overhead of 28 bytes per packet (UDP/IP Header overhead) this overhead is increased on a LAN by a further 12 bytes (Ethernet Header) or by 7 bytes over a PPP WAN link.
When transporting voice over low speed links (WANs) it is possible that normal data packets (e.g. 1500 byte IP packets) can prevent or delay the voice data from getting across the link. This can cause a very unacceptable speech quality. Thus it is vital that the routers in the network that carry voice have some form of Quality of Service mechanism (QoS). Version 2 supports the DiffServ (RFC 2474) Quality of Service mechanisms which are based upon the Type of Service (ToS) field in the IP header. The software will prioritise voice, fragment large packets and provide VoIP header compression to minimise the WAN overhead. Typically the VoIP WAN overhead is 35 bytes on 20 byte payload this is 175% overhead! On the WAN protocol this is reduced to only 6 bytes (3 bytes data , 2 bytes CRC and 1 byte HDLC flag) for the same 20 byte packet. The overhead is thus only 30% - a saving of 145%. When calculating the actual link speeds required to support voice traffic you must remember to include the aforementioned overhead. For example an 8Kbps compression voice channel actually requires 10.4Kbps of WAN bandwidth when using QoS or 22Kbps if using standard non QoS routers. QoS routers are also required to ensure low speech latency and to maintain sufficient audible quality. Therefore care should be taken when using third party routers to connect the two networks together.
From the above information it can therefore be deduced that VoIP functionality can be provided by the following:
If VCMs are fitted on the units at both ends then the voice can be encoded at 8k(G729), 6k3(G723), 8k(Netcoder) and 64k(A or u law), otherwise the voice will be encoded at 64k(A or u law). In the first instance the link between the two units can be 64k bps, or greater. The latter requires a link that is greater than 64k bps.
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