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Thanks, wardmundy. We have put most of the effort into Android version, but we also consider other platforms, such as Windows and iOS. But these are rather long-term plans.

We now have the Message Waiting Indication in Sipnetic. It works well with Asterisk and other PBXs supporting RFC 3842.

Not yet. We consider adding this feature in the next releases.

We have released a new version which allows calling normal phone from Sipnetic’s call history. Just long tap on the item and choose “Call via cell network”.

Currently Sipnetic doesn't use the Android's native Call Log, but we consider adding this feature. It'll take some time though. What we can do quickly is to add an option to make a normal phone call from Sipnetic's Log. I think it might be useful in your case.

We've updated the app. Now in-call screen has a button to show the DialPad. And it does have the # key.

Thanks. We'll add dialpad to the in-call screen in the next release.

Thanks for your feedback. The missing # key was a serious flaw on our part. We'll be fixing it ASAP. During the call you can press the button with lock icon to see codec and other call information.

Thanks for trying out our app. You should use Account Settings, not Network Settings. In the domain field type 1.2.3.4 (your PBX address without port), then set "Outbound Proxy" to "Specify Address" and "Proxy Server Address" to 1.2.3.4:your_port_number. As for Network Settings, they are used...

DPI can create troubles for RTP and SRTP, but not the SIP signaling itself, provided that you use TLS. In the Web world DPI is almost dead, since everyone has switched to HTTPS. I’d say there’s no reason not to use VoIP encryption. As for RTP media, there is a last resort option to run it over...

NAT traversal techniques are basically same for IAX and RTP. It’s either “Hosted NAT Traversal” aka “Latching” or some relay protocol like TURN. TURN + ICE (RFC 8445) is usually considered the best option, this is what WebRTC uses.

You mean DPI that simply blocks the unwanted traffic? Yes, SRTP is easily detected, but not decrypted without breaking TLS. DPIs that try to analyze or manipulate user’s traffic don’t make much sense in 2018, almost everything is encrypted. SIP signaling can use port 443, which is...

I don’t know if IAX can run over TLS, but with SIP you can use TLS+SRTP, which helps against ISP surveillance. ZRTP (RFC 6189) is even better, since it offers true end-to-end encryption. I have no idea how BLF function might look in a mobile messenger app, it’s more a hardware phone feature to...

Why? From my experience Asterisk works great with SIP.

Hi! We're the developers of a new VoIP app, Sipnetic. If you use an Android SIP client, such as CSipSimple, Zoiper, or Linphone, you should check out Sipnetic. I'll be glad to provide any info about this app or SIP protocol in general. Google Play...