Complete software solution on RISC-only SOC's for VoIP endpoint products
Highly optimized in CPU and memory utilization, vPort provides an SOC with substantial advantages in cost and power over multi-core or multi-chip VoIP products. It also simplifies software development, support, and maintenance because all of the software operates under a single operating environment using one set of tools. Initial applications for vPort include cost sensitive products such as Analog Telephone Adaptors (ATA), residential voice gateways, IP/Ethernet/WiFi phones and other VoIP-enabled consumer products.
Designed for portability, vPort can execute on both RISC and CISC platforms under standard operating systems, such as LINUX, and real-time operating systems. It is currently available on both ARM and MIPS based architectures under the VxWorks and LINUX operating systems. Because of its modular vOpen architecture, vPort can be quickly modified for customer-specific applications and runtime platforms.
vPort software is available out-of-the-box on a number of ARM and MIPS based reference hardware platforms for several target applications such as ATA, WiFi phones, and home routers. The software package includes reference application software, voice signaling protocols (SIP, MGCP, or H.323), packetizing (RTP, Jitter Buffer), voice processing (echo cancellation, voice compression, DTMF, etc.) and reference hardware-specific drivers (for audio and/or telephony devices).
Features
- Data, Security and Provisioning
- TCP (RFC 793)
- IPv4 (RFC 791)
- UDP (RFC 768)
- ICMP (RFC 792)
- ARP (RFC 826)
- DHCP Client (RFC 2131)
- DNS Client
- Static/next hop routing
- SNTP
- PPPoE
- Keep Alive
- STUN (RFC 3489)
- UPnP IGD
- DHCP Server
- NAT
- DNS Relay
- QoS Layer 2, 3
- Remote upgrade of software
- DoS Protection
- Web based management
- Advanced Telephony
- Caller ID
- Full Distributed Unicast Conferencing
- Call forwarding
- Call Return
- Call Blocking
- Do not Disturb
- Call Waiting
- Call Waiting/ Caller ID
- Speed Dial (Dial Plan)
- Multiple Line Appearances
- Distinctive Ring
- Re-Dial
- Internal calls
- Voice and Voice Signaling
- G711-Alaw; G711-MuLaw, G729AB and g.726 Voice Compression
- G.168 16 ms tail Echo Cancellation
- DTMF detection
- DTMF and Call Progress Tone Generation
- Packet Loss Compensation
- Silence Compression per G.711 Appendix I
- Fax pass-through (G711)
- Fax/Modem Tone Detection
- Caller ID Generation (FXS) US, China, Japan & ETSI
- Call Waiting
- Call Waiting Caller ID
- Custom Ring Cadences
- Gain Control (Volume adjust)
- DTMF Relay
- ITU T.38 Fax Relay
- SIP RFC 3261 & 1543
- RTP & RTCP RFC 1889 & 1890
- SDP
- Acoustic Echo Cancellation
- Full Duplex Speakerphone
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voip
- Voice Over IP Gateway Codec
- Voice and Video Engine
- SIP Analog Modem Server (Fully software V series data modems over VoIP)
- Multi-channel H.264 1080p video conferencing solution
- VOCAL’s T.38 Fax over IP (FoIP) software supports transmission of real time Group 3 fax/facsimile over IP networks
- G.723.1 ITU-T compatible vocoder