A Fax Analyzer is a media gateway appliance for debug and analysis of fax connections and transmissions over PSTN and IP networks. The analyzer provides at least one IP port to access IP networks and one Foreign Exchange Port (FXS) port to access the PSTN. The IP port captures session and Fax over IP data packets as well as provides access for remote control and monitoring of the appliance from another site. The FXS port may captures audio from a fax transmission over PSTN. Monitor and debug functions may be performed for T.38 real-time Fax over IP transfers or off-line using previously recorded files.
During a T.38 Fax Transfer, data sent from an analog fax machine is collected by a T.38 gateway and sent across the IP network in T.38 packets which can be transferred using TPKT over TCP, UDPTL over UDP, or RTP over UDP. The session control and T.38 packet stream is recorded to a file for processing. By using the sequence numbers in each packet, along with any redundant or forward error correction (FEC) packets sent, the full T.38 stream can be reconstructed. The T.30 dialogue contained within the T.38 stream is processed to reproduce the fax image.
For Fax over PSTN, fax demodulation software extracts images from a recorded file containing the intercepted PSTN audio signals. The facsimile monitor includes a fax decoder for both standard faxes as well as machines that make use of manufacturer specific T.30 ‘Non-Standard Facilities’ (NSF) features. VOCAL provides extensive software libraries to support capture and image extraction for debug and analysis for a variety of fax standards and transmission protocols.
- Capture V.17 and V.34 T.30 audio signals (PCM)
- Capture session control, Fax over IP, and T.38 packets
- Analyze T.30 fax transmissions using G3 and Super G3 protocols
- Analyze contents and timing of fax messaging and control sequences
- Compare message contents and timing for multiple Fax over PSTN or Fax over IP connections
- Decode fax images, including partial fax image recovery
- Playback Fax over PSTN
- Playback Fax over IP messaging for G.711 Fax Pass-Through and T.38 protocols
- Remote interactive debug and analysis
- Simple installation using local support technicians
- Quick identification and debug of fax transmission issues
- Cost effective use of remote engineering expertise
Simultaneous Calling using Single ATA
An Analog Telephone Adapter (ATA) with two FXS (RJ-11) ports enables multiple analog telephone devices (e.g. two phones or a phone and facsimile (fax) machine) to use Voice over IP (VoIP) technologies. Separate profiles are maintained to characterize the type and features of the VoIP session that can be supported for each device. Typically the ATA is assigned a phone number with a Universal Resource Identifier (URI) but it is possible to assign a different URI to each line (FXS).
When a call is received or placed, the Session Initiation Protocol (SIP) and the Session Description Protocol (SDP) are used to establish a connection between the two end points. When connected, audio signals are transmitted between the end points using Real-Time Transport Protocol (RTP).
An ATA configured with a single URI allows simultaneous phone calls using the same phone number. The data streams for the different calling sessions are directed to the corresponding ATA connected device. If a phone is connected to each FXS port, the ATA can ring the second phone when another call is received and the first phone is in use. If a phone and fax are connected to the FXS ports, the ATA can determine the type of session to establish for a received call and ring the phone or fax machine, as appropriate. Alternatively, a user could send a fax while talking on the phone.
An ATA configured with a URI for each line allows simultaneous phone calls using different phone numbers. For example, a user could have separate local and long distance phone numbers and possibly different service providers. In this case, the ATA directs the separate data streams to the phone number of the corresponding ATA connected device.
- Connect two analog telephone devices to VoIP services with one ATA
- Assign a single phone number to all analog devices
- Assign separate phone numbers to each analog device
- Make simultaneous calls using the same phone number
- Lower costs by maximizing calls over single phone number
- Make simultaneous calls using different phone numbers and calling plans
- Lower costs by using separate internet based calling plans
Media Gateway Access to PSTN and VoIP Services
An Analog Telephone Adapter (ATA) with one Foreign Exchange Subscriber (FXS) and one Foreign Exchange Office (FXO) port enables analog telephone equipment to make and receive telephone calls on both the internet using Voice over IP (VoIP) technology and on the Public Switched Telephone Network (PSTN). This allows the ATA device to fail over to the PSTN during power failures and 911 emergencies and the user to select from different calling services, e.g. PSTN for local and VoIP for long distance, to reduce telephony access costs.
In addition, this ATA configuration could be used as a media gateway to connect geographically remote offices using low cost internet VoIP services as well as access a local PSTN calling area from a remote office. In the same manner, a customer could dial the local PSTN number and connect through the ATA media gateway to someone in a remote office.
When a user dials a telephone number, the Session Initiation Protocol (SIP) services determine if the number is reachable directly over the internet or by PSTN. If the number is internet accessible, SIP would establish a voice connection using the Uniform Resource Identifier (URI). Otherwise, for a 7-digit number, it would automatically establish the connection to access the local PSTN through the FXO port. For a 10-digit number that is not internet accessible, SIP would be used to determine the remote office located in or nearest to the area code and then establish a connection to the number through the remote ATA FXO port to its local PSTN.
- Connect existing analog telephone equipment to both VoIP services and local PSTN
- Connect remote offices using media gateways and VoIP technology
- Use low cost internet VoIP services for long distance and PSTN for local calling
- Select from alternate paths for lowest cost voice connections
Access Multiple Networks
Analog telephone equipment can access networks with different capabilities through a Analog Telephone Adapter (ATA) media gateway. The media gateway provides 2 Ethernet (RJ-45) to connect to the different networks and 1 FXS (RJ-11) port to connect analog telephone equipment. The media gateway software supports the different protocols and/or codecs necessary to establish a connection to a destination on either network as well as convert data packets received from a network and transmit the audio signals to the analog device.
For example, an analog fax is connected to the FXS while the RJ-45 ports are connected to a LAN and WAN. The LAN uses the G.711 codec and Fax pass-through with RTP, while the WAN uses G.729A and T.38 Fax over IP with UDPTL for better compression, reliability, and reduced bandwidth utilization. When a call is dialed, the Session Initiation Protocol (SIP) and Session Description Protocol (SDP) establish a network connection with the type and characteristics appropriate for the two end-points. Once the connection is established, the fax transmission proceeds using the appropriate protocol and codec for that network. VOCAL’s extensive software libraries provide a variety of network protocols and codecs to send and receive fax and data over IP networks.
- Connect analog telephone equipment to multiple networks
- Support multiple protocols and codecs
- Use low cost internet services for multiple Voice and Fax over IP connections
- Select from alternate paths for lowest cost voice connections
Enable Analog Telephone Access to VoIP Services
An Analog Telephone Adapter (ATA) with 1 Foreign Exchange Subscriber (FXS) port enables an analog phone to make and receive calls on the internet using Voice over IP (VoIP) technologies. The ATA bridges the analog and digital domains by digitizing outgoing analog signals received from the phone and transmitting the data packets to the internet. Data packets from the internet are converted to analog signals and transmitted to the phone.
An ATA with one FXS (RJ-11) port connects an analog device (also multiple phones or fax and phone(s) connected to the same port) to the internet under a single phone number and Universal Resource Identifier (URI). A device profile identifies the type and characteristics for a calling session. When a call is received or placed, the Session Initiation Protocol (SIP) and Session Description Protocol (SDP) are used to establish a connection between the URI and the other end point. The Real-Time Transport Protocol (RTP) is used to transmit audio signals between the two end points.
If a fax is to be transmitted, the user dials a phone number on the analog fax unit and SIP/SDP establish a voice connection between the ATA and the destination end point. When the fax transmission is initiated, the ATA software can autodetect the fax modem audio signals and continue the fax transmission using G.711 encoded audio signals with VoIP pass through or the T.38 real-time Fax over IP protocol.
If a second call is received while a call is in progress, the ATA signals the user that a new call is pending. The user may choose to put the current call on hold and accept the new call or ignore the call. If the user accepts the call, the current call is placed on hold and a second session is established to connect the user to the new caller. Once the call is completed, the user can use the appropriate phone feature to resume the first call. If the user does nothing, a device busy response is transmitted after a specified time and the VoIP server can transfer the call to a voice mail service (if enabled) to record a message.
- Connect one or more analog fax/telephone devices to internet
- Call sessions using SIP, RTP/RTCP
- Fax and voice auto recognition
- T.38 Fax over IP
- Access expanded features of VoIP services from single ATA
- Send fax or call using the same line
- Lower telephony access costs with internet calling plans
Access VoIP Services with Legacy Secure Telephone Equipment
The secure Analog Telephone Adapter (ATA) provides a cost effective method for owners of analog Secure Telephone Equipment (STE) to connect their legacy devices to the internet. The ATA digitizes analog signals and transmits the packets using internet protocols to allow telephones, faxes and other analog devices to exploit the features and additional security provided by Voice over IP (VoIP) technology.
A secure ATA is configured with one Ethernet and one FXS port to connect a single STE to the internet. The adapter software suite provides for full Session Initiation Protocol (SIP) compliance and maximum connectivity with Network and VoIP protocols, voice codecs, as well as NAT/Firewall and Fax support. It also supports a web interface for remote updates and provisioning. VOCAL provides extensive software libraries with the necessary network protocols and codecs to enable analog STE access to internet Voice and Fax over IP services.
- Connect analog STE to internet VoIP services
- Interoperable with multiple VoIP providers
- Remote administration, configuration, and firmware updates
- Migrate legacy analog STE to IP networks
- Cost effective for large numbers of legacy devices
- Access enhanced features of VoIP technology
IP-PBX or Hosted VoIP
An IP PBX is a telephone system for business that transmits voice and video over a data network and interoperates with a PSTN. The system provides an internal LAN as well as a broadband connection to an external WAN or internet. A business can connect IP enabled devices directly or existing analog telephone and fax equipment via an analog telephone adapter (ATA) to the IP-PBX. This allows the company to manage services and features such as extension dialing, call forwarding, voice mail and auto attendant in-house to suit their business needs. Local, long distance and other advanced services can be accessed through the provider.
An internet private branch exchange (PBX) also called hosted Voice over IP (VoIP) is a distributed telephone service where most of the VoIP service equipment and services are located with the provider who also maintains the equipment. Companies are responsible for IP phones and other devices which connect through a router to access the advanced VoIP services of the provider. Small companies may decide to pay for the service rather than hire dedicated personnel to do the maintenance and provisioning in-house.
Both the IP-PBX and hosted VoIP systems provide access to lower cost VoIP service features. The IP-PBX provides local control of internal VoIP services although additional personnel will be required to support the equipment. On the other hand, the hosted VoIP system minimizes local equipment requirements but all hosted VoIP services are accessed through the provider. In this case businesses incur additional expenses for using the hosted VoIP services but avoid the costs of additional in-house equipment, maintenance and provisioning using their employees.
- Access VoIP services using IP-PBX
- Connect directly to hosted VoIP services
- Use lower cost VoIP services for long distance and local calling
- Control in-house VoIP services in-house with IP-PBX
- Avoid costs for in-house IP-PBX equipment and maintenance
Send T.38 Real-Time Fax over IP with Analog Fax Machine
An Analog Telephone Adapter (ATA) using T.38 real-time Fax over IP enables an analog Facsimile (Fax) machine to transmit a fax in real-time to an internet Fax service, another T.38 fax device, or analog fax unit connected to a T.38 capable ATA. Although the fax could be transmitted using a Voice over IP (VoIP) pass-through of G.711 encoded audio signals, T.38 real-time Fax over IP (FoIP) uses lower bandwidth and provides a more reliable method of fax transmission over the internet.
When a fax is sent or received, a voice connection is established between the ATA and the destination end point. A device profile identifies the type and characteristics for the calling session that is established. If desired, the device profile can be configured to always establish a T.38 connection for a dedicated fax device. Otherwise, when the fax transmission is initiated, on send the ATA software can autodetect the local modem tones or on receive autodetect a network event or T.38 RTP packet and initiate a T.38 session to complete the fax transfer.
- Supports Group 3 facsimile processing
- Supports both TCP/IP and/or UDP/IP transmission modes
- Configurable profiles support different Fax devices and features
- Configurable packet delay, jitter, redundancy for UDP/IP service
- Configurable forward error correction for UDPTL messaging
- Use end-to-end or local TCF generation for high speed training
- Access low cost internet Fax services
- Transmit fax using reliable T.38 Fax over IP using existing analog fax device
- Transmit fax to internet Fax service, T.38 fax, or other analog fax machine
SIP enabled ATA
An Analog Telephone Adapter (ATA) allows an analog phone to make and receive telephone calls on the internet using Voice over IP (VoIP) technologies. A Session Initiation Protocol (SIP) enabled ATA makes an analog phone connected to the adapter look and act like a SIP phone to VoIP service providers. The device profile for the unit connected to the adapter identifies the VoIP features it can support and is used with the Session Description Protocol (SDP) to setup the type and characteristics of the connection to establish.
SIP is used to initiate, setup, and terminate a session. To establish a connection between user agents, a SIP INVITE is issued by the sender and passed through a proxy to the invitee. It identifies the type and characteristics of the proposed connection. The invitee responds with 200 OK back through the proxy indicating the compatible options to use for this session. If the SIP ATA is protected by a router, the Simple Traversal of UDP through NAT (STUN) is used to establish voice connections through the NAT/firewall. Once the connection is established, the voice data is transmitted using Real-Time-Protocol (RTP) which readily passes through the NAT/firewall. To terminate the connection, the call initiator issues a BYE, a 200 OK is returned by the invitee, and the session is ended.
- Connect analog telephone equipment to internet VoIP services
- Access VoIP services through NAT/firewall
- Initiate, setup, and terminate calling sessions using SIP
- Use lower cost internet VoIP services
- Use expanded features of VoIP services with analog telephones
An Analog Telephone Adapter (ATA) Media Gateway can be used for diarization of conference call discussions. The gateway also connects an analog phone for a user to make and receive calls on the internet using Voice over IP (VoIP) services and participate in conference calling sessions. The gateway connects to all participants in a calling session and records the connection information and individual voice streams for each participant.
The Gateway provides an Ethernet port and a Foreign Exchange Subscriber (FXS) port to connect an analog phone to the internet using a single phone number and Universal Resource Identifier (URI). A device profile identifies the type and characteristics for a calling session. When a call is received or placed, the Session Initiation Protocol (SIP) and Session Description Protocol (SDP) are used to establish a connection between the URI and the other end point. The Real-Time Transport Protocol (RTP) is used to transmit audio signals between the two end points.
In a conference call, the SIP receives an Invite to join and SIP with SDP establishes the connections to each active participant or as they join the calling session. The voice streams for each participant are tagged with the source end-point and are recorded to either local mass storage or an NAS. When the recorded conference call is accessed following the meeting, the individual voice streams can be reconstructed using the source tags to associate voice packets with each participant. With conference diarization, the user can review and listen to the discussions as they occurred, by individual speaker, or as conversational threads. VOCAL provides extensive software libraries with support for protocols and codecs to create applications for conference calling and diarization.
- Connect analog telephones to internet VoIP services and access conference call sessions
- Record conference call discussions
- Maintain voice record of discussions
- Review conference by individual speaker, discussion thread, or as a whole.
Integrated Voice Data Router
Analog telephone equipment and personal computers can access IP networks using an integrated voice data router Analog Telephone Adapter (ATA). Typical routers provide 5 Ethernet ports: one to connect the router to an external WAN through a cable or DSL modem and four to connect computers, IP phones, or other network capable devices to a private LAN. Providing a Foreign Exchange Service (FXS) port (RJ-11) allows analog telephones, faxes and other devices access to Voice over IP (VoIP) and Fax over IP (FoIP) services, thus integrating voice and data services within a single device.
The router provides basic routing, Media Access Control (MAC) addressing, and security services. It employs Network Address Translation (NAT) and a firewall to protect the private LAN from external attacks. VoIP services use Session Initiation Protocol (SIP) and Simple Traversal of UDP through NAT (STUN) to establish voice connections through the NAT/firewall and Real Time Protocol (RTP) to transmit the voice data. VoIP applications rely on packet delivery in real-time, i.e. both delayed packets and lost packets can distort audio output. Therefore voice or RTP traffic has priority over data traffic. To achieve voice QoS levels, the router processes RTP packets ahead of other data packets.
- Connect analog telephone equipment to internet VoIP services
- Connect PCs and network devices to a private LAN
- Support NAT/firewall protection
- Support voice and data access to external WAN
- Simple access to voice and data internet services in one device
- Single lower cost alternative to using separate voice and data connection d