Golang builds sip service

With the popularity of VoIP technology, Session Initiation Protocol (SIP) has become a popular communication protocol. Using SIP, developers can easily build real-time communication applications such as phone calls, video conferencing, and instant messaging. As a fast, efficient and easy-to-use language, Golang provides good support for the construction of SIP services. This article will introduce how to use Golang to build SIP services.

1. Understand the SIP protocol

Before officially starting to develop SIP services, we need to understand the SIP protocol. SIP is a signaling protocol used to control signaling and media exchange in instant messaging sessions. It uses URLs to identify users and resources and provides a flexible way to establish, modify and terminate sessions.

SIP messages usually consist of two parts: request and response. Request messages are sent from the client to the server to request the server to perform some action, such as establishing or terminating a session. The response message is the server's response to the request, which includes a status code and possibly data or resources.

2. Install the SIP library

Before using Golang to develop SIP services, we need to use a SIP library to process SIP messages. Here we choose the pion/sip library. It can be installed through the following command:

go get github.com/pion/sip/v2

3. Create a SIP server

First, we need to create a SIP server to listen for incoming connection requests and SIP messages. Here is a simple example:

package main

import (
    "fmt"
    "net"
    "time"

    "github.com/pion/sip/v2"
    "github.com/pion/sip/v2/header"
    "github.com/pion/sip/v2/message"
)

func main() {
    host := "127.0.0.1"
    port := 5060

    // 监听UDP连接
    addr, _ := net.ResolveUDPAddr("udp", fmt.Sprintf("%s:%d", host, port))
    conn, _ := net.ListenUDP("udp", addr)

    transportLayer := sip.NewPacketConnTransportLayer(conn)
    messageInterceptor := sip.MessageInterceptor{}

    config := sip.Config{
        Headers: header.NewHeaders(),
    }

    handler := sip.HandlerFunc(func(writer sip.ResponseWriter, request *sip.Request) {
        switch request.Method {
        case sip.INVITE:
            // 处理INVITE请求，例如建立新的会话
            break
        case sip.REGISTER:
            // 处理REGISTER请求，例如注册新的用户
            break
        default:
            // 处理其他请求，例如ACK、BYE、CANCEL等
            break
        }

        // 回复200 OK响应
        okResp := message.NewResponseFromRequest(request.RequestURI, sip.StatusOK, "OK", config)
        writer.Write(okResp)

    })

    server := sip.Server{
        Handler:           handler,
        TransportLayer:    transportLayer,
        MessageInterceptor: messageInterceptor,
    }

    fmt.Printf("SIP server listening on %s:%d...
", host, port)

    err := server.Serve()
    if err != nil {
        fmt.Printf("Error starting SIP server: %s", err)
    }
}

In the above example, we first listen for UDP connections. Then create the transport layer object through the sip.NewPacketConnTransportLayer() function in the pion/sip library. This will be responsible for receiving and sending SIP messages. Next, we create a sip.MessageInterceptor object. It allows us to intercept and modify incoming and outgoing SIP messages. It is worth noting that we are not using interceptors here.

Next, we define a sip.HandlerFunc processing function to handle all incoming SIP requests. Here, we use switch statement to process based on the request type. For INVITE requests, we can implement our custom SIP logic here, such as establishing a new session. For REGISTER requests, we can register new users. Finally, we reply with a 200 OK response.

Finally, we pass the processing function to the sip.Server object. Then start the server and wait for SIP messages. In actual development, we may need to use more advanced configuration options to manage the server, such as TCP support, TLS support, Websocket support, etc. For specific options, please view the documentation of the pion/sip library.

4. Test the SIP server

Now we have created the SIP server and implemented the basic processing logic in it. We can use any SIP client application to test it. Below are some commonly used SIP client applications.

1. Linphone

Linphone is a free and open source SIP client application. It runs on multiple platforms, including Android, iOS, Windows, and Mac OS. You can download the corresponding version from the official website and install it.

After the installation is complete, we need to configure a SIP account to connect to our SIP server. In Linphone, you can add a new SIP account through the "Account" option in the "Settings" menu. Here, we need to specify the IP address and port number of the SIP server, username and password. After completing the configuration, we can click the "Register" button to connect to the SIP server and make calls.

2. Zoiper

Zoiper is another popular SIP client application. It offers many advanced features such as background mode, audio and video calling, file transfer, and more. You can download the corresponding version from the official website and install it.

Similar to Linphone, we need to configure a SIP account in Zoiper to connect to our SIP server. In Zoiper, you can add a new SIP account through the "Accounts" tab. Here, we need to specify the IP address and port number of the SIP server, username and password. After completing the configuration, we can click the "Connect" button to connect to the SIP server and make a call.

3. SIPp

SIPp is a popular command line SIP testing tool. It can simulate SIP clients and servers and conduct performance testing and load testing. You can download the corresponding version from the official website and install it.

When using SIPp, we need to write a SIP XML scenario file to define the SIP protocol and data used. The following is a simple SIP XML scenario example:

<?xml version="1.0" encoding="utf-8"?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic">
  <send retrans="500" retr_num="5" sip="sip:127.0.0.1" rport="5060">
    <![CDATA[
      INVITE sip:testUser@127.0.0.1:5060 SIP/2.0
      Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK123456789
      From: sip:testUser@127.0.0.1:5060;tag=123456789
      To: sip:testUser@127.0.0.1:5060
      Call-ID: 1234567890
      CSeq: 1 INVITE
      Contact: sip:testUser@127.0.0.1:5060

      Content-Type: application/sdp
      Content-Length: [len]

      [body]
    ]]>
  </send>
</scenario>

In this example, we define a scenario called "Basic". We use the send tag to send a SIP INVITE request to 127.0.0.1:5060. We also specify the message headers and SDP payload that need to be sent. In the SDP payload, we can specify details such as codecs, bandwidth, etc. for audio and video.

We can start the SIPp test through the following command:

sipp -sf <SIP_XML_SCENE_FILE> -s <SIP_SERVER_IP>:<SIP_SERVER_PORT> -run

In this way, we can start a SIP test in the command line, simulate a large number of requests and conduct performance and load testing.

5. Summary

In this article, we introduce how to use Golang to build a SIP server. We used the pion/sip library to handle SIP messages and created a simple SIP handler to handle all incoming requests. We also introduced some commonly used SIP client applications and command line testing tools. Through these tools we can test and optimize our SIP services for better performance and customer experience.

The above is the detailed content of Golang builds sip service. For more information, please follow other related articles on the PHP Chinese website!