Lock-free coroutine-based implementation of sip and rtp protocols. Hosted on jitpack repository https://jitpack.io/#soft-stech/sip4k.
Supported features:
- SIP and RTP protocol
- g711 codec for ip-telephony (asterisk for example)
- Multiple working clients on one host simultaneously. Designed for server application development.
Example of using:
@ExperimentalCoroutinesApi
fun main(args: Array<String>) {
val stream = FileInputStream("input.pcm")
GlobalScope.launch {
val client = Client(
sipProperties = SipProperties(
serverHost = "sipHost", // remote sip host
serverSipPort = 5060, // remote sip port
clientSipPort = 30200, // local sip port
login = "client", // your sip id
password = "password", // password
portsRange = Pair(40000, 40000) // range of rtp ports
),
rtpStreamEvent = { user, data ->
print("continue\n") // listener for processing stream rtp data in 16-pcm format
},
rtpDisconnectEvent = { user ->
print("disconnect\n") // callback for disconnect event
}
incomingCallEvent = { user ->
print("incoming call from $user")
}
)
client.start() // start sip client on local port
client.startCall("sipAbonent") // start one connection and open rtp stream
stream.use {
while (it.available() > 0) {
client.sendAudioData("sipAbonent", it.readNBytes(320)) // send piece of data in format 16-pcm having 20mc size
delay(20) // do 20mc delay since rtp protocol works only with UDP-header
}
}
}
runBlocking {
while (true) {
}
}
print("stopping!!!")
}Incoming call handling
client = Client(
incomingCallEvent = { user ->
GlobalScope.launch {
val stream =
FileInputStream("path/to/your16-pcm.file")
stream.use {
try {
while (it.available() > 0) {
client.sendAudioData(
user,
it.readNBytes(320)
) //send piece of data in format 16-pcm having 20mc size
delay(20) //do 20mc delay since rtp protocol works only with UDP-header
}
} catch (ex: Exception) {
throw ex
}
}
}
},
)