Cisco SPA122 Summary

The Cisco SPA122 ATA with Router combines VoIP services with an internal router for LAN connectivity. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections.

The Cisco SPA122 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. It also includes two 100BASE-T RJ-45 Ethernet ports for WAN and LAN connectivity. Each phone line can be configured independently. With the Cisco SPA122, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP voice with an extremely affordable, reliable solution.

Compact in design and compatible with international voice and data standards, the Cisco SPA122 can be used with residential, home-office, and small business VoIP service offerings, including full-featured hosted or open source IP PBX environments. This easy-to-use solution delivers advanced features to better connect employees and serve customers, all on a highly secure Cisco network.

• Enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection
• Provides high-quality, clear-sounding voice, using advanced voice quality-of-service (QoS) capabilities and the industry-leading voice Session Initiation Protocol (SIP) stack
• Supports reliable faxing with simultaneous voice and data use
• Includes two standard telephone ports, each with an independent phone number, for use with fax machines or analog phone devices, and one fast Ethernet WAN port, and one fast Ethernet LAN port for local home for business network connection
• Is compatible with all industry voice and data standards and common telephone features such as caller ID, call waiting, and voicemail
• Includes a simple-to-use web-based configuration utility for easy deployment

Cisco SPA122 Core Features

• Toll-quality voice and carrier-class feature support: The Cisco SPA122 delivers clear, high-quality voice
  communication under a variety of network conditions. Excellent voice quality in challenging, changeable IP
  network environments is made possible through the advanced implementation of standard voice coding
  algorithms. The Cisco SPA122 is interoperable with common telephony equipment such as fax, voicemail,
  private branch exchanges (PBXs) and key telephone systems (KTSs), and interactive voice response
  systems.
• Large-scale deployment and management: The Cisco SPA122 enables service providers to provide
  customized services to their subscribers. It can be remotely provisioned and supports dynamic, in-service
  software upgrades. A highly secure profile upload saves providers the time and expense of managing and
  preconfiguring or reconfiguring customer premises equipment (CPE) for deployment.
• Outstanding security: The Cisco SPA122 supports highly secure, encryption-based methods for
  communication, provisioning, and servicing.
• Compact size: Designed for small spaces, the Cisco SPA122 can be installed as a desktop unit or
  mounted on a wall.
• Comprehensive feature set: The standards-based Cisco SPA122 is compatible with Internet VoIP
  provider features such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more
  to provide a complete, affordable, and highly reliable VoIP solution.
• Easy installation and changes: The web-based configuration utility enables quick deployment and easy
  changes.
• Investment protection: Businesses that are growing rapidly can use the solution with other Cisco Unified
  Communications solutions, providing industry-leading investment protection.
• Peace of mind: Cisco solutions deliver the solid reliability you expect from Cisco. All solution components
  have been rigorously tested to help ensure easy setup, interoperability, and performance.

Cisco SPA122 Technical Specifications

• Data networking
  • MAC address (IEEE 802.3)
  • IPv4 (RFC 791) upgradeable to IPv6 (RFC 1883)
  • Address Resolution Protocol (ARP)
  • Domain Name System (DNS) A record (RFC 1706) and SRV record (RFC 2782)
  • Dynamic Host Configuration Protocol (DHCP) server and client (RFC 2131)
  • DHCP client reservation
  • DHCP Options 159 and 160
  • Point-to-Point Protocol over Ethernet (PPoE) client (RFC 2516)
  • Internet Control Message Protocol (ICMP) (RFC 792)
  • TCP (RFC 793)
  • User Datagram Protocol (UDP) (RFC 768)
  • Real Time Protocol (RTP) (RFC 1889) (RFC 1890)
  • Real Time Control Protocol (RTCP) (RFC 1889)
  • Differentiated Services (DiffServ) (RFC 2475) and type of service (ToS) (RFC 791/1349)
  • VLAN tagging (IEEE 802.1p)
  • Simple Network Time Protocol (SNTP) (RFC 2030)
  • Upload data rate limiting: static and automatic
  • QoS voice packet prioritization over other packet types
  • MAC address cloning
  • Port forwarding
  • SIP channels support both UDP and TCP transport
  • VPN pass-through with IP Security encapsulating security payload (IPsec ESP), Point-to-Point Tunneling
  • Protocol (PPTP), and Layer 2 Tunneling Protocol (L2TP)
• Voice gateway
  • SIPv2 (RFC 3261, 3262, 3263, and 3264)
  • SIP proxy redundancy: dynamic through use of DNS SRV A records
  • Reregistration with primary SIP proxy server
  • SIP support in Network Address Translation (NAT) networks (including Serial Tunnel [STUN])
  • Highly secure (encrypted) calling using Secure RTP (SRTP)
  • Codec name assignment
  • G.711 (A-law and μ-law)
  • G.726 (32 kbps)
  • G.729 A
  • Dynamic payload
  • Adjustable audio frames per packet
  • Dual-tone multifrequency (DTMF): in-band and out-of-band (RFC 2833) (SIP information)
• Voice features
  • QoS (Ethernet port upstream bandwidth control)
  • Independent configurable dial plans with interdigit timers and IP dialing (1 per port)
  • Call progress tone generation
  • Jitter buffer: Adaptive
  • Frame loss concealment
  • Full-duplex audio
  • Echo cancellation (G.165 and G.168)
  • Voice activity detection (VAD)
  • Silence suppression
  • Comfort noise generation (CNG)
  • Attenuation and gain adjustments
  • Flash hook timer
  • Message waiting indicator (MWI) tones
  • Visual messaging waiting indicator (VMWI) using frequency shift keying (FSK)
  • Polarity control
  • Hook flash event signaling
  • Caller ID generation (name and number): Bellcore, DTMF, and European Telecommunications Standards
  • Institute (ETSI)
  • Streaming audio server: Up to 10 sessions
  • Music on hold
  • Call waiting, call waiting and caller ID
  • Caller ID with name and number
  • Caller ID blocking
  • Selective and anonymous call rejection
  • Call forwarding: No answer, busy, and all
  • Do not disturb
  • Call transfer, call return, and call back on busy
  • Three-way conference calling with local mixing
  • Per-call authentication and associated routing
  • Call blocking with toll restriction
  • Distinctive ringing: Calling and called number
  • Off-hook warning tone
  • Advanced inbound and outbound call routing
  • Hotline and warmline calling
  • Long silence (configurable time setting) silence threshold
  • Disconnect tone (for example, reorder tone)
  • Configurable ring frequency
  • Ring validation time setting
  • Tip and ring voltage adjustment setting
  • Ring indication delay setting
• Fax capability
  • Fax tone detection pass-through
  • Fax pass-through using G.711
  • Real-time fax over IP using T.38 fax relay (T.38 support is dependent on fax machine and network and transport resilience)
• Security
  • Password-protected system reset to factory default
  • Password-protected administrator and user access authority
  • Provisioning, configuration, and authentication
  • HTTPS with factory-installed client certificate
  • HTTP digest: Encrypted authentication using MD5 (RFC 1321)
  • Up to 256-bit Advanced Encryption Standard (AES) encryption
  • SIP Transport Layer Security (TLS)