i am creating an app for internet calling using VoIP . But have no idea how to send SIP registration to sever to stabilise connection between to devices .
and any suggestions for the library
You can use Linphone library for that purpose. Although be prepared that documentation is not ideal and you will need to figure some of the things by yourself. You however can inspire by this tutorial project
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Currently building a mobile application that ideally would have functionality that allows one user to "nudge" another user, pretty much the equivalent to a Facebook "poke." For the sake of building quickly, we'd like to use as many third-party libraries/services as possible. So far we were looking into using pusher and expo, but to be totally honest, we are a bit confused as to how to architecture this.
From what I understand, the front-end (ios app) would subscribe (open a socket) to a specific channel/server and the server would then send messages to the client(s) that subscribed whenever necessary. How this works with users being subscribed to every other user they are friends with, I don't understand. What's the proper way to architecture this?
I also read that on the front end, when a user doesn't have the ios application open you can't have any websockets open to the server, so how does that work?
Additionally, how does one bundle notifications??
Really confused with this, any help would be greatly appreciated.
Chris
Well, push notifications now mostly are implemented with Firebase. This is a service inside most of mobiles, so you only need use these libraries to implement your client application.
From the server, you only need to consume a rest api of Firebase.
Now there are permissions to allow mobile applications receive push notifications in background. You do not need to implement a socket or websocket, that is a feature of Firebase.
Read more about this.
https://medium.com/google-cloud/push-notification-for-react-native-bef05ea4d1d0
https://dzone.com/articles/how-to-add-push-notifications-on-firebase-cloud-me
I have build a PJSIP app for VoIP.
Which works well when trying to connect a client which is using the same network.
I am maintaining the connection over TLS.
I have successfully configured RTPEnginge
And using Kamailio 4.3
I have tried all the methods on the PJSIP help site for getting through NAT.
I am able to connect the call with a client but unable to hear any audio.
Please help or suggest a way to solve this issue.
Any help or guidance is much appreciated.
The exact reason could certainly be confirmed only by getting much more informations.
However, from your description, you may be simply missing a RTP relay: kamailio usually requires rtpproxy to be installed.
You will also need to configure kamailio (kamailio.cfg) to use rtpproxy.
Here is the documentation for kamailio's rtpproxy module.
Is there any WebRTC solution for iOS for free with easy setup?
I tried to use http://www.webrtc.org/native-code/ios because our web end is already done with its web api and I thought I may not have other way around for letting calls go between web and iOS too. But iOS API's setup is very tedious and time taking (The downloading of WebRTC checkout is taking like lives with no gain).
I searched around and found a few like tokBox and quickblox but they are not free.
Did you look at RestComm iOS SDK ? It supports WebRTC Audio only right now but we are working on adding video in the next few weeks. Also it uses SIP as a signalling protocol.
https://github.com/Mobicents/restcomm-ios-sdk
http://www.telestax.com/how-to-integrate-the-restcomm-ios-client-sdk-in-your-app/
http://docs.telestax.com/restcomm-client-ios-sdk-quick-start/
Take a look at https://github.com/oney/RCTWebRTCDemo .
This is a React Native WebRTC project which works on iOS and Android and also has a signaling server example (but you can also use the online version for quick tests!).
Since the WebRTC requires DTLS-RTP, RTCP-FB, ICE and a lot of other newest standards, but the VoIP standards are old about 10+ years, therefore you need setup a gateway to convert the signaling and transcoding the RTP.
With the WebRTC Gateway, in the browser side, you can create the HTML5 application to connect to WebRTC gateway, the gateway will communicates with your PBX, and your iOS client connects to your PBX, then the call can be established between browser with iOS client app.
I am trying to implement Asterisk server for VOIP in an iPhone application.
I am able to register the user through manually, but I wanted to do the same thing programmatically. I am not sure but there might be some idea like sending through an HTTP request.
If you have any idea or pointer please share with me.
Thanks
I think the thing that you are looking for is called SIP client. You install the client on the iPhone and it will register to your SIP server (let it be Asterix or anything else). Then you can make VoIP calls between the clients registered contemporary, send instant messages, exchange files or even implement some custom protocol over SIP to exchange other type of data.
There are a lot of SIP client applications for iPhone, and even a couple of SIP stacks that is a framework that allows you to build your own SIP client application through its API calls, so you will have full control on everything. One of the most mature of these with iPhone support is pjsip. Check out their web site to see whether this is what you are looking for.
it is hard because is necessary modify the dial plan in order to add a sip account. but you can program a web service that read a text file with the extensions from your mobile.
it is nto fully clear from your desription what exactly you want, but probably this one will help
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
I have to implement VOIP in iPhone app,which is the best way of performing this.
If you have any idea please share with me or give me some pointer from where i can explore and get some better solution.
Thanks
Yes you can have iPhone Voip app. Could you please let me know what will be the purpose or functionality do you want to have in this app.
If you want to make calls using VoIP uses SIP(session initiation protocol) for communication to VoIP tor VoIP to PSTN. Follow the below URL for iPhone VoIP Client.
http://code.google.com/p/siphon
Get the source code and compiled it using the instructions from below URL;
http://code.google.com/p/siphon/wiki/Compilation
For VoIP Stack Please visit www.pjsip.org.
Use pjsip. It's a SIP library written in C. pjsip uses GPL license.
You may find documentation to do so.