sip

1）创建服务 编辑gwp-svc.yaml apiVersion: v1 kind: Service metadata: name: gwp spec: ports: - port: 80 targetPort: 8080 selector: app: gwp 创建服务 kubectl create -f gwp-svc.yaml 查看服务 kubectl get svc 在运行的容器中远程执行命令，其中两个双斜杠（--）代表着kubectl命令项的结束，之后时pod内部需要执行的命令 kubectl exec gwp-fgjg6 -- curl -s http://10.98.20.174 同一个服务可以暴露多个端口，此时需要指定端口名称 编辑gwp-svc.yaml apiVersion: v1 kind: Service metadata: name: gwp spec: ports: - name: http port: 80 targetPort: 8080 - name: https port: 443 targetPort: 8443 selector: app: gwp 此外，targetPort还可以使用命名端口（就是pod中配置的containerPort对应的名称），最大的好处是即使更换端口号也无须更改服务spec。 2）服务发现 通过环境变量发现服务

I am wondering how you can get the call id when you make an outgoing call using the android.net.sip API. I am currently just making an outgoing call as they do in the android sip demo. call = manager.makeAudioCall(me.getUriString(), sipAddress, listener, 30); I also saw in the documentation that you can create a sip session when making a call in order to get the call id, but I just can't figure it out. See http://developer.android.com/reference/android/net/sip/SipManager.html#createSipSession(android.net.sip.SipProfile for the documentation on the SipManager . I am also doing this before I

I was reading about SIP and found that there's an java API JAIN SIP to develop SIP based applications. Then I also found that there is JAIN SLEE and SIP servlets. I believe JSLEE and SIP Servlets are containers to deploy applications based on JAIN and SIP servlets respectively. Can some please explain. Also, please tell me which one is better to start with for learning. Thanks JAIN SIP is a java specification ( https://jcp.org/en/jsr/detail?id=32 ) for writing Java SIP applications in a standard and portable manner (between JAIN SIP vendors). JAIN SLEE is a java specification ( https://jcp.org

I am still learning about SIP and all its protocols, specifically trying to integrate PJSIP into an iPhone application to make p2p calls. I have a question about a peer 2 peer connection using PJSUA. I am able to make calls perfectly to other clients on my local network by calling directly using the URI: sip:192. . .*:5060 I am curious if this will work for making direct calls to other SIP URIs that are not on the local network without using server configuration - if not this way, is there another way of making p2p calls without server configuration? thanks in advance, You can make calls

Apple SIP简介及在Clover中如何控制 来源 http://www.yekki.me/apple-sip-overview-and-how-to-disable-it-in-clover/ 什么是Apple SIP Apple SIP（System Integrity Protection）机制是OSX 10.11开始启用的一套关键的安全保护技术体系。 SIP技术的整个体系主要分为： 文件系统保护（Filesystem protection） 对于系统文件通过沙盒限制root权限，比如：就算你有root根限，也无法往/usr/bin目录写入。 运行时保护（Runtime protection） 受保护的关键系统进程在运行状态下无法被代码注入，挂调试器调试，以及限制内核调试等 内核扩展签名（Kext signing） 10.10中强制要求签名，要想绕过这个限制，就必需加入启动参数“kext-dev-mode=1”（10.11 DB5开始，”rootless=0”的启动参数也被废除了），这个启动参数在10.11中被废除。另外，10.11官方要求第三方kext必须被安装至/Library/Extensions。 Apple官方如何对SIP保护技术进行配置 进入10.11的安装程序或Recovery HD 使用其中所带的终端进行相关操作。在此环境下，由于特殊启动标志位的存在

So I'm using the native sip library, and I can connect and register with the server just fine. And when I make the call, it hits a proxy that routes it to a regular phone call, then calls the number inputed. It will connect fine, and the phone on the other end receives the call, but there is no audio. I know the proxy can handle audio because there is an iPhone app hitting the same server and it connects just fine. Here's my code for making the call : public void makeCall(String s) { SipAudioCall.Listener listener = new SipAudioCall.Listener() { @Override public void onCallEstablished