VoIP Integration in App & Web

Question

I have a very general question on how to implement VoIP for our current mobile & Web App. (we have an Android+iOS App and a Web Application based on AngularJS/NodeJS).

What we want to achieve

In the first step we want to achieve inter Application Voice and Video Calls. Later on we might expand into outbound calls into the normal telephone network. But this post is mainly for getting info on how to implement only our first step.

general thoughts

We had some experiences with Asterisk before which turned out to be far from easy. So for this project we wanted to get some feedback before actually implementing anything.

thoughts on technology

At first I thought it might be a good idea to use WebRTC, but since it's only supported on Chrome, FF and Opera for the moment and pretty much is unsupported for native mobile Apps we think that WebRTC is probably out of the picture for now. (or do you think otherwise?) After searching the web a bit more we found this: http://www.webrtc.org/native-code

Has anyone experience with this libs? It seems to us, that this could be the best solution for a modern voip solution (and also would allow us to skip the asterisk server)

The second idea would be to setup an Asterisk Server for ourselves. Every time a user logs into the App we would connect him as a SIP Client to the asterisk. If one user calls the other one we think we should be able to make the call for example with the node package Asterisk Manager API (https://github.com/pipobscure/NodeJS-AsteriskManager).

The third idea would be to use a SIP Provider, but at the moment I'm not sure if that's really the best idea.

Since we're no VoIP experts, are there any other possibilities for VoIP integration into our apps?

Any thoughts on that subject would be very appreciated! Thank you!

Answer 1

The main factor is the network configuration that you app will be working with. Given you're using mobile clients and web apps it's almost certain that you're using the internet and also likely that you'll have 3G and 4G mobile networks in the mix (3G/4G cause a lot more problems for VoIP than WiFi).

Given the above assumption holds the biggest challenge your app will have is establishing media (audio and/or video) connections between mobile clients which are behind different NATs and in a lot of cases multiple NATs. There is almost no chance you'd be able to get by without a server here. The server will be needed to act as a relay point for the media streams for the mobile clients. You will use the RTP protocol for the media and working out how to get it reliably from client A to client B is your biggest obstacle. The signalling side - whether it be SIP, web sockets or something else - will be secondary (note both SIP and WebRTC use RTP to carry the media).

If I were in your shoes the steps I'd take would be:

1. Install and try out some softphones (blink, bria, zoiper et al) on your own mobile devices, find a SIP provider that supports video calls and get some experience with calls. It may not be the experience you anticipated...

2. Once you are comfortable with the softphone experience you would then need to make two decisions:

   • Whether to deploy your own server or use an existing provider,

   • Whether to write your own client, find an existing one or something in between.

I can answer the deploy your own server question. You don't want to do that unless the VoIP part of your app is going to be something you charge for and make a good margin off. Running a VoIP server and all the security and network considerations that go along with it is a full time job. It may start out being easy but once a few customers start connecting and the fraudsters come along it will take on a life of its own. In the decade I've been messing around with SIP I'd estimate 75% of providers have gone out of business and it was their full time job.

Besides all that I'd be surprised if there wasn't a SIP provider that suited your needs. These days there are highly sophisticated services available that led you control every aspect of your call flow with your own code (anveo, tropo, twilio) right down to free services (sip2sip, sipbroker) that may be all you need to get started.

For the client software there are various SIP SDK's you'll be able to leverage (pjsip).