What is the difference between the SPA2100 and the SPA2102?
In short, the SPA2102 is the upgraded version of the SPA2100. Therefore its box looks slightly different to the older model.
The main difference is the upgraded port. The SPA2100 has a 10mbps port, while the SPA2102 has the 100mbps port. In most applications, the 10mbps port is more than efficent for standard VoIP functions.
As a side note, the firmware for the SPA2102 will not work on the SPA2100. So make sure you always update your configuration according to the proper model.
Below is a datasheet comparison between the Linksys SPA2100 and the SPA2102.
Physical Interfaces Comparison
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SPA2100
- Two RJ-45 10base-T Ethernet ports
- Two RJ-11 FXS Ports for analog circuit telephone device (Tip/Ring)
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SPA2102
- Two RJ-45 100base-T Ethernet ports
- Two RJ-11 FXS Ports for analog circuit telephone device (Tip/Ring)
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Voice Gateway
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SPA2100
- SIPv2 – Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264)
- SIP Proxy Redundancy – Dynamic via DNS SRV, A Records
- Re-registration with Primary SIP Proxy Server
- SIP Support in Network Address Translation Networks – NAT (incl. STUN)
- Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
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SPA2102
- 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])
- Secure (encrypted) calling via prestandard implementation of Secure RTP
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Voice Specifications
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SPA2100
- Dynamic Payload
- Adjustable Audio Frames Per Packet
- Fax Tone Detection Pass-Through
- Real Time Fax via T.38 (Pending)
- Voice Band Data Modem Support (Pending)
- DTMF: In-band & Out-of band (RFC 2833) (SIP INFO)
- Call Progress Tone Generation
- Jitter Buffer – Adaptive
- Frame Loss Concealment
- Full Duplex Audio
- Echo Cancellation – (G.165/G.168)
- VAD – Voice Activity Detection w/ Silence Suppression
- Attenuation / Gain Adjuustments
- Flash Hook Timer
- MWI – Message Waiting
- Indicator Tones
- VMWI – Via FSK
- Polarity Control
- Hook Flash Event Signaling
- Caller ID Generation (Name & Number) – Bellcore, DTMF, ETSI
- Music on Hold Client
- Streaming Audio Server – Up to 10 Sessions
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SPA2102
- Dynamic payload
- Adjustable audio frames per packet
- 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 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.)
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Caller ID (CID)
The "called from" name and number that is transmitted and displayed over telephone lines. The caller has the ability to block delivery of this information by dialing 67 before dialing the destination number.
Dial Plan
A set of rules used by SIP devices to determine what digits are actually sent to the switch and what number are restricted. The simple Linksys PAP2 and SPA2100 have a single dial plan string that you can edit to allow 7 digit dialing in your local area and block 1-900 calls.
DTMF (Dual Tone Multi Frequency)
When using a touch tone dial pad you generate a sound made up of two separate tones at different frequencies. The Telephone Company recognizes these tones and dials the appropriate number or code. Supposedly the human voice cannot produce dual tones at different frequencies which is why AT&T adopted this standard; it prevents you from causing the phone to dial while you speak.
Gateway
A device that connects to the network and converts a signal to be transmitted via IP. In telephony the classic use of a gateway is to convert POTS or analog lines to IP. As an example, we use a gateway to bring SIP Trunks into an xBlue Key System.
Hold
A business feature where a call is temporarily held by the KSU or Server and the caller typically hears music or an announcement during this time. Hold is used while sending the caller to another extension, while setting up a conference call, or while answering another call or simply when looking up information.
IP (Internet Protocol)
A protocol that specifies the way data is broken into packets and the way those packets are addressed for transmission. Unlike traditional telephone lines which create a straight path from caller to recipient; IP uses the best available path at any given moment to move data packets between point A & point B. The beauty of IP is that the sending end and receiving end communicate success in sending and receiving to insure all data packets are delivered. A really great concept of IP is that of self healing. If a route is broken and the data packets cannot be delivered an alternate route is automatically selected.
Server
The central processor in a network. In our case the server supports the telephony functions in a VoIP network. It manages the connections between the phone lines and the telephones, voicemail, etc.
SIP (Session Initiation Protocol)
Internationally recognized IP telephony signaling protocol used for VoIP. This is the most widely used protocol in the market and is a standard meaning anybody who builds a SIP 2.0 compliant product should work with any other SIP 2.0 compliant device. For instance our Talkswitch is compliant so any off the shelf SIP phone will work as an endpoint for a home phone or a remote worker. The trick with SIP 2.0 is that the combined devices will work to the feature set of the lowest common denominator meaning a SIP 2.0 telephone only support 80% of the SIP feature set then even though the telephone system supports a higher set of features, you will only have the base features of the phone.
SIP Proxy
Acts as the gatekeeper in a SIP connection. For instance the SIP Proxy for connection to a SIP trunk holds the translation information necessary to make the connection between your telephone system and the SIP trunk provider.
VoIP (Voice over Internet Protocol)
This is simply the carrying of voice via IP meaning a standard voice stream is broken into packets at the transmit end, sent over a standard data network (such as the Internet) and reassembled into a voice stream at the receiving end. VoIP traffic is trickier than normal data traffic because the timing of packets on the receiving end is critical. For more detail see Basics of the Internet.
