Skip to end of metadata
Go to start of metadata

Extended Microsoft UC Features (VXe)

VX 4.7.1 enhances VX to support Microsoft's Office communication server SIP directly. In order to support direct connection the following enhancements have been made to VX.

  • Microsoft RT-Audio codec
  • Polycom G.722.1 @16K (SIREN) codec
  • Interactive Connectivity Establishment (ICE) draft 6
  • Forward error correction via Redundant RTP (RFC2198)
  • Scalable Secure RTP
  • Microsoft Quality of Experience server integration
  • Direct configuration into Office communications environment
  • Microsoft Media Relay Authentication service
  • Dynamic codec selection and network detection

Prior to these features the mediation server role is used to convert these advanced pieces of the Microsoft Office Communication server network to base SIP. 

With these enhancements VX can now talk directly to the Microsoft Office Communications Server front-end role. The audio path is directly from VX to Microsoft Office Communicator or the required end-point. This removes the need for the mediation server role in the network.

The VX 4.7.1 implementation of the Extended UC features has minor limitations that will be removed in subsequent releases. Please refer to section 3.4 for limitation details.

Redundant Audio Date in RTP Payload

This feature adds support for Redundant Audio Data in RTP payload (RFC 2198) which is negotiated using SDP for a SIP call.

Media Port Change Restriction

This feature allows the user to configure the RTP port range, which is used in SIP and H323 call. The permissible range is 16384-32767 and the first port must be an even number. An alarm is generated when a free port is not available for new call.

MRAS Support

This feature supports a PSTN call traversing through VX and destined to a client sitting outside the Enterprise Network. In this case, the audio stream through VX traverses through the TURN/STUN server. This feature provides the authentication credentials which VX uses to traverse through the TURN/STUN server. VX enables interaction with an A/V Edge Authentication Server, also known as a Media Relay Authentication Server (MRAS).

Multiple MLine SDP

This feature provides a way to transport multipart body contents in one and same SIP message.

RT Audio

The RT Audio codec support feature includes support for the SIP and BSP signaling and its negotiation in VX, including the hardware compression and decompression on the STIX card.

RFC3960 Support for SIP to SIP Calls

This feature supports RFC3960 for local ringing tone generation for SIP to SIP calls.

QoE Server Support

This feature enables VX to interact with the QoE server to report call metrics and statistics. The report is sent as an XML report embedded in a SIP SERVICE message body.

TURN Client Support

This feature enables VX to setup keyholes for the audio stream through the TURN server in order to relay its media stream through the TURN server.

SSRTP

This feature provides confidentiality, message authentication and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP); and improves performance when the same RTP payload is distributed to hundreds of recipients.

New CAS Modes and Permanent Circuits

VX 4.7.1 is enhanced to support an own number on each CAS channel. The own number will be used for the calling number for any calls from the channel.

VX is also enhanced to support different CAS types on each timeslot. This allows mixing of MRD, E&M and other CAS protocols.

This feature includes CAS Tunnel and CAS MRD Protocol.

  • CAS Tunnel implementation allows tunneling of signaling and voice from CAS lines in a VX node to CAS lines in another VX node. Thus there is a permanently open duplex path between the trunks in both the nodes.
  • CAS Manual Ring Down protocol (CAS MRD) protocol interworks with SIP. This enables VX to act as a gateway between a SIP PBX and a TDM PBX running the MRD protocol. Once the call is established, VX maintains the call in a connected state unless the call has been disconnected from the SIP PBX.

Support for VX behind a NAT

When a user on the internal network initiates a request to the Internet, their private IP address is translated to the public IP address when the request goes through the firewall/Network Address Translation (NAT) to the Internet. This is so the destination site knows the IP address to which it should return the information. The firewall/NAT maintains a log of which endpoints requested what destinations, and when a response is received from a destination site, the firewall/NAT directs it to the endpoint that made the original request.

When the VX is on a private network and needs to establish a call to a SIP UA in the public network, the call goes through a NAT server. The NAT translates the IP address in the IP header, but generally, it is not able to translate the IP Address in the SIP and SDP Headers.

This feature enables VX's signaling and audio stream NAT traversal assuming that the VX is placed behind a NAT on the Private Network side. The VX uses the Public IP of the NAT behind which it is placed, in all SIP and SDP messages for making SIP calls to/receiving SIP calls from devices on the public internet.

One number fax

The issue with a single number solution including fax is the user experience.  The fax experience has always been the same:  place fax into machine, dial number and press GO.  In a single number solution, both phone and fax calls reach an end-user's phone; they both end up ringing the number's owner.  If the call is a fax, the user must then hang up and wait for the fax to retry.  During the retry, the phone must not be answered so that the fax can be delivered to the recipient's UM mailbox.  When the phone rings again, the recipient cannot know whether it is the fax retrying or a real person-to-person call.  The fax will continue to retry the recipient until the phone is ring-no answer, or until the recipient manually forwards the call to his UM mailbox.  The single number fax experience is confusing for both sender and recipient.

In VX 4.7.1 this enhancement provides a true single number Unified Messaging solution for voice, voicemail and fax. The One Number Fax feature enhances VX to re-route a call when a fax tone is detected. This allows the call to be re-routed from OCS to the UM server so the fax can be delivered.

The One Number Fax feature enhances VX to re-route a call when a fax tone is detected. This allows the call to be re-routed from OCS to the UM server so the fax can be delivered.

VLAN for UDP

This feature is an enhancement of the handling of UDP packets. Configuration changes include the two tables for DiffServ Mapping and 802.1p Priority are replaced with one combined table called Packet Priority Table. This table allows for up to eight (8) pre-set combinations of 802.1p and DiffServ, to be used for all inbound and outbound traffic.

RFC3960 Support for SIP to SIP Calls

RFC3960 is a technique to generate ringback for SIP calls. SIP does not have a standard way of determining inband vs. local ringback. This is further complicated with a call is forked. Forked calls can have mixed local and inband ringback. RFC3960 removes this ambiguity by monitoring the RTP stream. Local ringback is played after a 180 is received and no RTP stream exists. As soon as RTP is detected the local ringback is stopped and the audio is cut through. VX has supported RFC3960 since release 4.4.4 for ISDN to SIP calls.

SIP to SIP calls did not support RFC3960; this can cause no ringback with certain IP-PBX's. While it is always possible to configure around this and provide ringback this was often as the expense of inband ringback.

4.7.1 Implements full RFC3960 ringback generation for SIP to SIP calls.

E1 MLPP

Multi Level Precedence and Preemption support allows government, especially military, users to be able to classify the priority of a call so that a high priority call can override lesser priority calls in the event of congestion of low resources. For example, an emergency call can override a conversational call when all lines are full.

MLPP provides a documented and standardized way to coordinate this system, and interoperate with other equipment that also provides MLPP functionality.

VX 4.7.1 adds E1 MLPP to allow VX to support MLPP on both T1 and E1 circuits.

VLAN and Diffserv configuration changes

VX System Release 4.7.1 introduces enhancements to the configuration of Virtual Local Area Network (VLAN) and Diffserv priorities. The new priority table greatly simplifies the configuration of VLAN and Diffserv on the system.

Enhancement to the Timer System

This feature is an enhancement to change the system time from the CLI and also helps change the system time by the time server. This feature is used every time the system time is changed either from the time server or manually from the CLI.

  • No labels