PCBEST NETWORK SIP Software Development Kit(SDK)
Configuration Items
gtsrv.sip.server.model
Application type.
1 = server (default).
0 = client phone.
gtsrv.sip.ip.address
If you have multipul ip address and you want your application only run on one of those ip address,
enable this option, and fill the ip address. Defaultly it will listen on all network interfaces.
gtsrv.sip.ip.port
Local sip ip port.
gtsrv.sip.rtpstartrange and gtsrv.sip.rtpendrange
RTP port range
gtsrv.sip.prefered.codec = 102,101,100,3,98,0,8
prefered audio codecs and order
18 = G729
102 = Speex
104 = iLBC 30ms
103 = iLBC 20ms
3 = GSM
98 = G726-32
0 = G711 - Mulaw
8 = G711 - Alaw
for example, if you want to only use mulaw and alaw, then set gtsrv.sip.prefered.codec
= 0,8
if you would like to use GSM, mulaw, and alaw, but GSM first, then please set gtsrv.sip.prefered.codec
= 3,0,8
gtsrv.licence.key
input your licence key here if you purchased one
gtsrv.sip.boardnum.per.server, gtsrv.sip.spannum.per.board, gtsrv.sip.channum.per.span
you can define how many boards on one server, how many spans on one board,
and how many channels on one span.
gtsrv.sip.boardnum.per.server * gtsrv.sip.spannum.per.board * gtsrv.sip.channum.per.span
is the total channels you have.
gtsrv.sip.boardnum.per.server can be 1-20.
gtsrv.sip.spannum.per.board can be 1-16.
gtsrv.sip.channum.per.span can be 1-30.
gtsrv.sip.reg.client.num
How many sip accounts you want to use to register with SIP server.
If it is greater than 0, then please give each account information by setting the
following items:
SIP account information. the following tags are for the first account, and if you have more than one account, please add reg2, reg3, ...
gtsrv.sip.reg1.displayname
Display name, can be any.
gtsrv.sip.reg1.username
User name. usually digits for sip provider.
gtsrv.sip.reg1.domain
SIP server domain name or ip address.
gtsrv.sip.reg1.proxy
SIP server acctually proxy domain name and ip address. same as domain name in most cases.
gtsrv.sip.reg1.authorization
Authorization name. usually same as user name.
gtsrv.sip.reg1.password
Password.
gtsrv.sip.reg1.expire
In how many seconds to register.
gtsrv.sip.reg1.register
If register on the server to receive incoming calls. 1 = register(Default). 0 = no
gtsrv.sip.use.nat.addr
Set it to 1 if the SIP proxy server is in local network.
gtsrv.sip.stun.server
he stun server to be used to discover global IP address. If you don't know
any, just leave it blank. SDK will use "stun.fwdnet.net" as default STUN server.
gtsrv.sip.callcontrol.auto.answercall
If it answers incoming calls automatically.
gtsrv.sip.callcontrol.auto.transfercall
If trasnfering is allowed.
gtsrv.sip.dxsound.device
The key word of your sound capture and playback device. You don't need
to set it usually if the default windows playback and capture device is the sound
hardware your want to use with ActiveX phone.
gtsrv.sip.dxsound.device.playback
The key word of your sound device for playing sound. If you want to use different sound device to playback, then set this value.
gtsrv.sip.dxsound.device.capture
The key word of your sound device for recording sound. If you want to use different sound device to record audio, then set this value.
gtphone.audio.record.enabled
If enable audio-record. This tag can be changed dynamically when application running, so softphone can dynamically open recording according to users' choice.
1 = enable audio-record.
0 = no audio-record
gtphone.audio.record.rootdir
Root folder for audio-record.
gtsrv.sip.dtmf.method
0 = Using DTMF inband audio
1 = SIP Info
2 = RTP, RFC 2833.
3 = Auto(RFC2833 or in audio) (Default)
"gtsrv.sip.vad.enabled"
If enable VAD(Voice Activity Detection). "1" = enable. "0" = no (Default)
"gtsrv.dx.play.agc.level" and "gtsrv.dx.capture.agc.level"
These two tags are for DirectX AGC(Automatic Gain Control) process. the agc level is the percent of max volume for the sound buffer.
The valid range is 0.5f to 1.0f, but the recomended range is 0.8f to 0.95f. At 1.0f, some clipping might be experienced. Sample: CFG_SetValue("gtsrv.dx.capture.agc.level", "0.85")
"gtsrv.sip.on.in.vad" and "gtsrv.sip.on.out.vad"
For VAD(Voice Activity Detection). Set it to 1 to enable. Defaultly it is disabled(0).
On_VoiceActivityDetected event will be triggered if it is on.
"gtsrv.sip.callcontrol.auto.acceptcall"
Defaultly it is 1, means accpet call automatically and send SIP TRY response once the call is in. If you set it to 0, then you have to use Send_Accept later to accept the call.
"gtsrv.sip.callcontrol.auto.ringcall"
Defaultly it is 1, means send SIP 180 RING response once the call is in. If you set it to 0, then you have to use Send_Ring later to ring the call.
"gtsrv.sip.on.rtp.packet"
Defaultly it is 0, means no accessing to rtp package. If it is 1, it means On_RecvRTPPacket and On_SentRTPPacket events will be triggered to access RTP package.
"gtsrv.sip.on.dx.audio"
Defaultly it is 0, means no accessing to directx audio stream. If it is 1, it means On_CaptureDXAudio and On_RenderDXAudio events will be triggered to access RTP package.
"gtsrv.sip.conference.room"
How many conference room to open. (Defaultly it is 0, no conference room)
"gtsrv.sip.on.in.vad"
If detect incoming voice activity. (Defaultly it is 0)