Research-Lab™ VOIP H323 DLL Soft Phone SDK 1.5 for Windows Development
VOIP H323DLL Soft Phone SDK brings H.323 protocol support for ActiveX. With this SDK one can create a simple program to connect and start speaking with anyone with a direct IP Address (provided no NAT Router or Firewall is set) or use the Gateway or the Gatekeeper to connect to /or from PSTN lines. Now VOIP open source projects have made it easier to implement a customized VOIP Solution eliminating costly PSTN lines. Though this implementation might turn out fairly complex, our VOIP Implementation Team at Research-Lab will guide you remotely for the same.
If you have a good quality Internet connection you can get your phone service delivered to you through your Internet connection instead of from your local phone company. We apply only one time nominal $399 charge for the latest version and Implementation Guidelines all for a download, see the sample below. Please note, this includes a PowerPoint Presentation for the best of H323 Soft Phone Implementation for your home or office.SDK Features and Applications:
- IP-Address based direct local connectivity within LAN
- NetMeeting Compatible
- Gateway Specification with/without Login
- Gatekeeper Specification with/without Login
- Proxy/Firewall support for Version 2.0 [i.e. for Registered Users]
- Voice/Ring Volume Adjustment [i.e. for Registered Users]
- Save a conversation to a wave file [i.e. for Registered Users]
- Create a Answering Machine
- Create a Interactive Voice Response with Speech Recognition in any language
- Create a Professional Call Center
TECHNICAL DETAILS
===============
Usage : RPhone [options] -l : RPhone [options] [-p host] hostname/alias
where: hostname/alias = Remote host/alias to call
Options:
-a --auto-answer : Automatically answer incoming calls
-d --autodial host : Autodial host if phone off hook
-h --help : Display this help message.
-l --listen : Only listen for incoming calls
-v --verbose n : Set amount of information displayed (0=none)
--disable-menu : Disable internal menu
--ringfile filename : Set sound file for "ring" annunciation
--ringdelay seconds : Set delay between playing above file
--save : Save parameters in configuration file.
Gatekeeper options:
-g --gatekeeper host : Specify gatekeeper host.
-G --gatekeeper-id name : Specify gatekeeper by ID.
-n --no-gatekeeper : Disable gatekeeper discovery.
-r --require-gatekeeper : Exit if gatekeeper discovery fails.
--password pwd : Password for gatekeeper H.235 authentication.
-p --proxy host : Proxy/Gateway hostname/ip address
Divert options:
-F --forward-always party : Forward to remote party.
-B --forward-busy party : Forward to remote party if busy.
-N --forward-no-answer party : Forward to remote party if no answer.
--answer-timeout time : Time in seconds till forward on no answer.
Protocol options:
-i --interface ipaddr : Select interface to bind to for incoming connections(default is all interfaces)
--listenport : Port to listen on for incoming connections (default 1720)
--no-listenport : No listen port
--connectport port : Port to connect to for outgoing connections (default 1720)
--connectring num : Distinctive ring number to send to remote - 0 (default) to 7
-b --bandwidth n : Limit bandwidth usage to (n * 100) bits/second
-f --fast-disable : Disable fast start
-T --h245tunneldisable : Disable H245 tunnelling.
-u --user name : Set local alias name(s) (defaults to login name)
-S --disable-h245-in-setup Disable H245 in setup
--tos n : Set IP Type of Service byte to n
--setup-param string : Arbitrary data to be put into H.225 Setup PDU
--portbase port : Base port for H.245 and RTP data
--portmax port : Maximum port for H.245 and RTP data
--translate ip : Set external IP address to ip if masQueraded
Audio options:
-e --silence : Disable silence detection for GSM and software G.711
-j --jitter [min-]max : Set minimum (optional) and maximum jitter buffer (in
milliseconds).
--recvol n : Set record volume
--playvol n : Set play volume
Video transmit options:
--videodevice dev : Select video capture device (default /dev/video0)
--videotransmit : Enable video transmission
--videolocal : Enable local video window
--videosize size : Sets size of transmitted video window size can be small (default) or large
--videoformat type : Set capture video format can be auto (default) pal or ntsc
--videocolorfmt format : Set the preferred capture device color format can be RGB24, RGB24F, RGB32, ...
--videoinput num : Select capture video input (default is 0)
--videotxquality n : Select sent video quality,(def 9). 1(best)<=n<=31
--videotxminquality n : Select video quality lower limit,(def 1). 1(best)<=n <=31 A value of 4 works best for NetMeeting
--videofill n : Select number of updated background blocks per frame 2(def)<=n<=99
--videotxfps n : Maximum number of video frames grabbed per sec 2<10(def)<30
--videosendfps n : Target minimum number of video frames sent per sec 0.001<6(def)<30
--videobitrate n : Enable bitrate control. 16< n <2048 kbit/s (net bw)
Video receive options:
--videoquality n : Set received video quality hint - 0 <= n <= 31
--videoreceive viddev : Receive video to following device
: null do nothing
: ppm create sequence of PPM files
Video options:
--videotest : Display local video. Exit after 10 seconds. NO h323 call
Sound card options:
-s --sound device : Select sound card input/output device
--sound-in device : Select sound card input device (overrides --sound)
--sound-out device : Select sound card output device (overrides --sound)
--sound-buffers n : Set sound buffer depth (default=2)
VoIPBlaster options:
-V --voipblaster num : Use device number
Audio Codec options:
-D --disable codec : Disable the specified codec (may be used multiple ti
mes)
-P --prefer codec : Prefer the specified codec (may be used multiple tim
es)
--g711frames count : Set the number G.711 frames in capabilities (default
30)
--gsmframes count : Set the number GSM frames in capabilities (default 4
)
--g7231 : Set G.723.1 as preferred codec
--gsm : Set GSM 06.10 as preferred codec (default)
--g711-ulaw : Set G.711 uLaw as preferred codec
--g711-alaw : Set G.711 ALaw as preferred codec
--g728 : Set G.728 as preferred codec
--g729 : Set G.729 as preferred codec
--g7231 : Set G.723.1 as preferred codec
-I --input-mode mode : Set the mode for sending User Input Indications (DTMF) can be string, signal, q931 or rfc2833 (default isstring)
-U --user-input-cap mode : Set the mode for User Input Capabilities can be string, signal, rfc2833 or none (default isall)
Retail Package Includes:
Free 1 MB Redistribution Silent Installer.6 More Client-Server Samples.
Proxy/Firewall support for Version 2.0 [i.e. for Registered Users]
Voice/Ring Volume Adjustment [i.e. for Registered Users]
Save a conversation to a wave file [i.e. for Registered Users]
Applications
- Create a Professional Quality Answering Machine
- Create a VOIP Answering Machine
- Create a Call Center for your company
More TAPI Telephony and Speech Recognition Samples Demoshttp://www.research-lab.com/newcomp.htm
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Contact tim@research-lab.com for additional information.