SIPvxml SIPvxml Columbia SIP-based VoiceXML browser
Architecture
Design overview
API documentation
SIP-based VoiceXML browser Session Initiation Protocol (SIP) is a signaling protocol used for establishing and terminating Internet telephony call. VoiceXML is a language designed to create audio dialogs that feature synthesized speech, digitized audio, recognition of spoken and DTMF key input and recording of audio for telephony applications. The SIPvxml module implements a SIP interface to our VoiceXML browser for interactive voice response applications to telephone users. It uses Columbia LIBSIP++ and RTPLIB++ libraries for SIP and RTP, respectively.
Main features include:
- Uses SIP for signaling and RTP for media transport
- Supports RFC2833 for signaling of DTMF digits (audio/telephone-event)
- Supports call transfer, both blind and bridged.
A user can call up a number and get transferred to a audio conference (addressed by the a SIP URI) or be transferred to another number.- Based on VoiceXML 1.0 specification
- Interworks with Cisco IP phone(7960).
- Supports service specification in SIP URI.
- Can accept input in SIP instant message also. Generates prompts for both audio and instant message.
Some of the planned features include: support for more tags and attributes as per VoiceXML 2.0 specification, support for RFC 2198 for multiple digits per packets, support for audio/tone type and speech recognition.
We have developed example VoiceXML applications using web CGI scripts written in Tcl, compatible with our SIPvxml tool for the following:
- Application to check voice mail. Users can call a phone number(which is mapped to the url which points to the sipvxml browser) and can enter their TPIN and access their voicemails.
- Application to check the volume level of the microphone and the speakers of your phone. This can be used by users before joining the conference. The application will help the user adjust the volume level of the microhone and the speakers.
- Telephonic interface to a conference user. Similar to the case of the voice mail access, a phone number which is mapped to sipvxml can be used to provide an menu driven telephonic interface to users which have restricted capabilities at their client(a POTS phone or a hardware phone) for authentication.
- Accessing email by phone that allows sending and fetching emails via IMAP. This is integrated in our CINEMA system and uses the email account configured via the CINEMA web interface.
- SIPquest developed an auto-attendant application for corporate front desk. It prompts the user with various phone numbers or user names to reach the user phone.
- A calling card kind of application that can allow using CINEMA account for authentication to make long distance or international calls.
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