sip4k

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
                        }
                    }
                }
            },
           
)