sip

How to Hide a SIP connection

由 别来无恙 提交于 2019-12-11 19:45:23
问题 I'm developing a SIP mobile softphone, customer needs a complete hiding of SIP messages from softphones to SIP servers as VOIP calls are regionally prohibited, however using TLS connection was not sufficient since the message headers are easily recognized as a SIP message. What are the best common alternative? what about openvpn, IPSec tunneling? 回答1: Transmitting SIP over TLS means the SIP headers will only be viewable if someone is able to compromise your TLS keys, i.e. it's highly unlikely

How to get call info from Kamailio

由 女生的网名这么多〃 提交于 2019-12-11 19:43:56
问题 I have setup a Kamailio server and am able to establish calls. I need a way to get call related information like from, to, duration,etc. I have enabled the dialog module in the config but no avail. I am not well versed with config files and I am not sure if I am doing something wrong in the config file. 回答1: You need to Modify the config file to log the call related information in kamailio database tables.Here's the link You have to uncomment the lines in the config file those add columns to

How to finish activity in broadcastreceiver onCallEnded()

由 亡梦爱人 提交于 2019-12-11 19:13:29
问题 I have a question, how to finish activity in brodcastreceiver onCallEnded (SIP) . There is a brodcastreceiver like this: public class IncomingCallReceiver extends BroadcastReceiver { @Override public void onReceive(Context context, Intent intent) { SipAudioCall incomingCall = null; try { SipAudioCall.Listener listener = new SipAudioCall.Listener() { ... @Override public void onCallEnded(SipAudioCall call) { // IncomingCallActivity.finish(); } }; Main mainActivity = (Main) context;

Android SipManager throwing NullPointException

由 回眸只為那壹抹淺笑 提交于 2019-12-11 18:29:18
问题 I'm trying to implement my own android sip implementation and I am having problems registering with my SIP server. Specifically, the SipManager.register() function is throwing a NullPointerException. Has anybody experienced this before? There isn't anything that is null in the below function so I don't understand where this is coming from. Thanks public boolean initializeProfile() { final Globals global = Globals.getInstance(); if (sipProfile != null) closeLocalProfile(); String username =

Restcomm Cloud throwing ERR_CONNECTION_TIMED_OUT when connecting via WSS://mydomain.com:5083/

由 ↘锁芯ラ 提交于 2019-12-11 15:17:12
问题 I have created my domain (XYZ.restcomm.com) on https://cloud.restcomm.com and i have created some Accounts and created RVD App and created short demos too. Now i am creating a Node JS based Application to open the WSS connection to my restcomm cloud instance by using https://github.com/RestComm/restcomm-web-sdk. And the code i am in my HTML Page is as follows I have ensured that i am able to access my restcomm cloud domain "XYZ.restcomm.com" and i created the account with user id "guest" and

PJSUA2 Android - Incoming calls drop after 32 seconds

由 天涯浪子 提交于 2019-12-11 15:09:50
问题 I'm building a PJSUA2 (PJSIP 2.8) Android app and I have some issues: i.e. only on incoming call, call state remains in "PJSIP_INV_STATE_CONNECTING" and after 32 seconds the call drops. I'm looking for the cause of the issue since several days, I googled a lot and all what I found is: in most situations this issue is related to NAT management or network issues related to NAT. In a few words: in most cases the called party does not receive the ACK after answering the call. Finally I was able

Android Sip incoming Call using Service with Broadcast Receiver

由 ￣綄美尐妖づ 提交于 2019-12-11 12:22:36
问题 Hai Every one Actually am trying to create an application which supports for making a audio call over the internet with SIP based,here am using native sip,and i was facing the problem with the incoming call,i have done the registration part with service,but while doing the incoming call i can not able to answer the call kindly please help me Service file package example.com.myapplication; import android.app.Notification; import android.app.NotificationManager; import android.app.PendingIntent

sip 180 183区别

由 淺唱寂寞╮ 提交于 2019-12-11 10:07:22
sip 180 183区别： 180 不带sdp, 183带sdp信息； 如果A的SIP终端收到183，它就协商媒体，将B端发过来的Early Media在自己的扬声器里放出来；但如果收到的是180，没有SDP就没法协商媒体，因此，B就没法给A发Early Media了。怎么办，总不能让主叫用户干等着啊，所以，A的话机在这种情况下能自己产生一个回铃音，或任何用户在A话机上设置的音乐 首先，我们先看一种熟悉的情况。FreeSWITCH可以假装它就是B，这样，配置方法跟上面讲的基本一样，只是它在假装后还要假戏真做，要用bridge这个Application再去呼叫B，并把电话接通。 <action application="ring_ready"/> <action application="sleep" data="2000"/> <action application="answer"/> <action application="playback" data="/tmp/hello.wav"/> <action application="bridge" data="user/B"/> 所以在上面的配置中，至于是回180还是183，配置方式跟上面讲的一模一样，就没必要多说了。 其次，FreeSWITCH心情好，想听听B的意见。如果它即不执行ring_ready，也不执行answer

How does apps like Whatsapp or telegram listen to the incoming call/message events on Android?

由 ≡放荡痞女 提交于 2019-12-11 07:05:40
问题 I built a VoIP calling app which maintains a persistent connection with the server to listen to any incoming calls. I implemented a background service to do this. But since Oreo, this running code is now broken because of the introduction of Background Execution Limits After looking into forums, I found that some people are suggesting Convert Service to JobService and let android schedule it Doing so, my app won't be able to receive calls when it is stopped Run your operations in foreground

Initiating call and receiving call in web browser using freeswitch

由 老子叫甜甜 提交于 2019-12-11 06:35:09
问题 I have a requirement, i have a web site in which i want to implement outgoing call and incoming call functionality. I am using freeswitch on windows as sip server, currently i am able to initiate call on local extensions using verto, how can i initiate outbound calls to mobile phone directly from browser and also able to receive the calls using browser. 回答1: Freeswitch has webrtc support, which means you can use SIP-webRTC client to register from browser and do IN/OUT calls. SIP-webRTC client
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