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VX supports access to standards-based Voice over IP (VoIP) networks through the Session Initiation Protocol (SIP) or the H.323 protocol. In addition, VX supports a proprietary protocol BSP for efficient node-to-node communication (see the VX VoIP Protocol Support and VX PSTN Protocol Support charts below for all currently supported protocols and associated interfaces).

SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions or calls. SIP can be used to initiate sessions as well as invite members to sessions that are advertised and established by other means. VX can terminate and originate calls to devices using the SIP protocol. SIP calls terminated on a VX can be routed over the PSTN using CAS E&M, CAS R2, ISDN, or SS7 or back over VoIP using SIP, H.323, or BSP protocols.

VX can terminate and originate calls to devices using H.323 signalling and converts the signalling to common PSTN protocols such as ISDN, SS7, or CAS E&M, CAS R2, or back over VoIP using SIP, H.323, or BSP protocols.

VX VoIP Protocol Support




ITU-T H.323v2


ITU-T H.225.0


ITU-T H.245




NET Proprietary

VX PSTN Protocol Support






Harris 20/20










Japan NTT




Loop Start










ITU-T White book


ITU-T Blue book





Subscriber Mode Using Loopstart CAS

Once a call gets connected to VX in subscriber mode, the exchange (phone or PBX) cannot ask the subscriber (VX) to go on hook. The disconnect request must be initiated only from the subscriber (VX).

Users can use a VX Script for numerous reasons when VX is in Subscriber mode, some examples are:

  • To indicate call disconnect by playing re-order
  • To indicate the other side is busy by playing a busy tone.

To do the above actions the VX script should handle the `clearback' condition.

If Clearback is handled, it must always end after a finite amount of time otherwise the call will continue indefinitely.


In the course of attempting to complete a call, a VX node can try to route a call over the IP network or try various CODECs. If the call cannot be completed, the VX node can be configured to route the call onto the circuit switched infrastructure, using defined routing digits, to complete the call as a hairpinned connection.

Hairpinning Example

Using the example network depicted above, hairpinning occurs as a result of the following sequence of events:

  1. User located at Phone A calls Phone B .
  2. The call is sent to the VX.
  3. The call cannot go through because the IP network is either down or congested.
  4. VX then hairpins the call over the PSTN to another operator (carrier), using configured, extra routing digits.

Network Address Translation Traversal

Network Address Translation (NAT) is a logical function typically configured in a border router that connects a public network to a private network. NAT translates IP addresses from packets that traverse the associated boundary. NAT simplifies and conserves IP address usage and enables private IP networks that use nonregistered IP addresses to connect to the Internet. NAT operates in two realms – private and public. Using NAT, the IP address(es) of the private network can remain hidden, which results in the following benefits:

  • Eliminate necessity to renumber the private network when providers change.
  • Allow a large address space inside the private network to be mapped to a smaller set of addresses on the outside, which provides privacy and a degree of security.

NAT has become a widely used tool for network address conservation and is particularly useful within home networks and small enterprises where the purchase of large public address space is impractical and costly.

Because NAT basically performs translation service on any TCP/UDP traffic that does not carry source and/or destination IP addresses in the application data stream or transport layers, application-layer protocols that include IP addresses and ports within the payloads\ – such as VoIP protocols (H.323 and SIP)--cannot operate using NAT.

VoIP protocols (H.323 and SIP) consist of signaling and media parts, both of which require endpoint IP addresses to enable communication. These addresses are normally embedded in the packet payload. When a VoIP packet traverses a typical NAT device, the NAT device translates the private IP address unchanged and therefore incorrectly passes it to the other end, which results in a VoIP connection failure. To resolve this NAT traversal issue, VX implements an algorithm that determines the correct IP addresses for RTP (real-time transport protocol) traffic to use in media connections.

SIP Signaling with NAT

The VX NAT Traversal feature uses an intelligent relay method for Session Initiation Protocol (SIP) calls, whereby SIP calls are routed through trunk groups. The SIP Inbound Call Routing option specifies the trunk-group to handle SIP messages, as well as the IP address range.

If a trunk-group is configured to work with NAT traversal, and the source address in the IP Header of SIP packet is different from the Via/Contact/Route Headers, that SIP packet is treated as coming from a SIP entity behind a NAT device. The VX node\ – acting as the SIP Registrar--handles the storing of the registered contact information, the contact address, the source address, and the port. VX enforces SIP registration renewal to keep bindings active. The renewal time is controlled by the TTL on the trunk-group configuration.

This feature does not work for calls from a public network to private network using TCP or TLS.

H.323 NAT Implementation

H.323 does not require any changes to its call control (signalling) to work through a NAT device for outbound calls. The same algorithm used with SIP applies to H.323 media connections for outbound calls.

SS7 Link Over Bearer Port

VX supports provisioning of SS7 links by channels within ISUP bearer ports. You can set any timeslot as an SS7 channel in an ISUP bearer port.


The SIP-HSI Module has been enhanced with the introduction of a new SIP transport layer that encapsulates the transport specific functions and provides TCP and UDP communication capabilities.

Authentication Proxy

VX can be configured to act in "Proxy-Like" mode to forward registrations that it receives from SIP phones. The SIP phones have to be configured with the VX address as their proxy for this function to operate. This feature allows pass through facility for PBX features like call hold, unhold, and call transfer.

H.245 Tunnelling

H.245 tunnelling conveys messages within a Q.931 call signalling channel instead of establishing a separate H.245 channel. This conserves resources, synchronizes call signalling, and control as well as setup time.