Cisco Systems SPA2102 manual Voice gateway, Voice algorithms, Voice specifications, Fax capability

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Data Sheet

Voice gateway

Session Initiation Protocol (SIP) v2 (RFC 3261, 3262, 3263, 3264)

 

SIP proxy redundancy-dynamic via 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 via prestandard implementation of Secure RTP

Voice algorithms

Codec name assignment

 

G.711 (A-law and -law)

 

G.726 (16/24/32/40 kbps)

 

G.729 A

 

G.723.1 (6.3 kbps, 5.3 kbps)

 

Dynamic payload

 

Adjustable audio frames per packet

Voice specifications

Dual-tone multifrequency (DTMF): in-band and out-of-band (RFC 2833) (SIP info)

 

Flexible dial plan support with interdigit timers and IP dialing

 

Call progress tone generation

 

Jitter buffer-adaptive

 

Frame loss concealment

 

Full duplex audio

 

Echo cancellation (G.165/G.168)

 

Voice activity detection (VAD) with silence suppression

 

Attenuation/gain adjustments

 

Flash hook timer

 

Message waiting indicator (MWI) tones

 

Visual message waiting indicator (VMWI) via frequency shift keying (FSK)

 

Polarity control

 

Hook flash event signaling

 

Caller ID generation (name and number)-Bellcore, DTMF, European

 

Telecommunications Standards Institute (ETSI)

 

Music on hold client

 

Streaming audio server-up to 10 sessions

Fax capability

Fax tone detection pass-through

 

Fax: G.711 pass-through or real-time fax over IP via T.38 (T.38 support is dependent on

 

fax machine and network/transport resilience.)

 

 

Security

Password-protected system reset to factory default

 

Password-protected administrator and user access authority

 

Provisioning/configuration/authentication:

 

HTTPS with factory-installed client certificate

 

 

HTTP digest-encrypted authentication via MD5 (RFC 1321)

 

Up to 256-bit Advanced Encryption Standard (AES) encryption

Provisioning, administration,

Web browser administration and configuration via integral web server

and maintenance

Telephone keypad configuration with interactive voice prompts

 

Automated provisioning and upgrade via HTTPS, HTTP, Trivial File Transfer Protocol

 

(TFTP)

 

Asynchronous notification of upgrade availability via NOTIFY

 

Nonintrusive, in-service upgrades

 

Report generation and event logging

 

Stats in BYE message

 

Syslog and debug server records-per-line configurable

 

Per-line and purpose configurable syslog and debug options

Physical interfaces

Two 100BASE-T RJ-45 Ethernet ports (IEEE 802.3)

 

2 RJ-11 FXS phone ports-for analog circuit telephone device (tip/ring)

Subscriber line interface

Ring waveform: trapezoidal and sinusoidal *

circuit (SLIC)

Maximum ringer load: 3 ringer equivalence numbers (RENs)

 

 

On-hook/off-hook characteristics:

 

On-hook voltage (tip/ring): -50V nominal

 

Off-hook current: 25 mA min

 

Terminating impedance: 8 configurable settings including North America 600 ohms,

 

European CTR21

© 2008 Cisco Systems, Inc. All rights reserved. This document is Cisco Public Information.

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Contents Product Overview HighlightsFeatures SpecificationsSpecifications Data networking Voice gateway Voice specificationsProvisioning, administration MaintenanceEnvironmental Dimensions Power supplyIndicator lights/LED Package ContentsC78-502137-00 11/08
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SPA2102 specifications

The Cisco Systems SPA2102 is a versatile Voice over Internet Protocol (VoIP) adapter that serves as a bridge between traditional telephony systems and modern IP networks. Designed primarily for small to medium businesses, the SPA2102 is highly regarded for its reliability, ease of use, and rich feature set. This device allows users to make and receive phone calls over the internet while maintaining the ability to connect traditional analog phones.

One of the standout features of the SPA2102 is its dual-port capability. The device includes two FXS ports, allowing users to connect two separate analog telephones. This makes it an ideal choice for businesses that want to retain their existing telephony infrastructure while transitioning to a VoIP system. The ability to utilize two telephone lines simultaneously provides flexibility and convenience, especially for users in a busy office environment.

The SPA2102 leverages Session Initiation Protocol (SIP) technology, which is widely recognized for its robustness and interoperability. The support for SIP ensures that the SPA2102 can work seamlessly with various VoIP service providers, offering users a broad range of options for their telecommunication needs. In addition to SIP, the device supports various codecs, including G.711, G.726, and G.729, allowing for flexible audio quality settings and bandwidth management.

Security is also a critical aspect of the SPA2102. It incorporates advanced encryption methods, such as Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS), to protect voice communications from potential eavesdropping. This focus on security makes the SPA2102 a reliable choice for businesses concerned about the confidentiality of their conversations.

The device is easy to configure and manage, thanks to its web-based interface. This allows administrators to quickly set up the adapter, manage network settings, and troubleshoot any issues that may arise. Furthermore, the SPA2102 supports Quality of Service (QoS) features, ensuring that voice traffic is prioritized over other types of network traffic, which enhances call quality and reliability.

Overall, the Cisco SPA2102 is a powerful, user-friendly VoIP adapter that combines traditional telephony with modern IP technology. Its dual-port capability, support for SIP, extensive security features, and ease of configuration make it a solid choice for businesses looking to innovate their communication systems while minimizing disruption. Whether used in a small office or a larger organizational setting, the SPA2102 continues to be a reliable component of VoIP solutions.