Matrix Telecom SARVAM UMG System Manual page 23

The key features of the VoIP Interface are:
• Upto 250 SIP Trunks - for Proxy or Peer-to-Peer (non-Proxy).
• 128 Maximum Simultaneous Voice Calls (as per License).
• Selectable Network Assignment (Connection Type) - Static IP, DHCP, PPPoE.
• Selectable DNS - Automatic and Static.
• Dynamic DNS for mobile SIP devices.
• STUN.
• TCP and UDP NAT Keep Alive.
• VLAN.
• Symmetric RTP Selection.
• MAC Address Cloning option.
• Fax over IP - T.38 (UDPTL), T.38 (RTP) and Pass Through.
• Send CLI Option for outgoing calls
• Selectable DTMF - RTP (RFC 2833), SIP Info, InBand
• Flash Detection using SIP INFO and RFC2833.
• Broad Voice Codec Selection -
G.711 (A-Law)
• Quality of Service - RTP DiffServe/ToS
• Voice Mail Subscription for SIP Trunks.
SIP Trunks
The SARVAM UMG application supports a maximum of 250 SIP Trunks, allowing you to subscribe to as many as
250 different Internet Telephony Service Providers (ITSP).
You can connect SARVAM UMG application to the IP network, which may be Public Internet.
The Single Line Telephone Interface
The Single Line Telephone (SLT) Interface allows any standard, two-wire, analog single line telephone
instrument —- rotary, pulse-tone, cordless, feature phones with or without Calling Line Identification — to be
connected to the SARVAM UMG as extension phone.
The SLT Interface has the following features:
• Selectable Caller ID Presentation - DTMF, FSK
• Programmable Ring Type - Trapezoidal, Sinusoidal, Low Trapezoidal, Low Sinusoidal.
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G.729, G.723, GSM FR, iLBC (30ms), iLBC (20ms), GSM EFR, G.711 (u-Law),
Matrix SARVAM UMG System Manual