(note: Only the call part is Vanilla SIP. The procedure for registering a Facetime user into their servers etc. is all non-SIP, encrypted/ciphered.)

(for my user review of the iphone4 and bumper read here)

Well heck, good job Apple! I just tested facetime and did a quick check on its protocol. No hacking needed – just an on the wire black box inspection – its just plain SIP and STUN for firewall discovery. Apple plans to make this protocol public, and they seem to have done an excellent job. And thanks for showing the world that you don’t need complicated encryption and proprietary tunneling tricks for an excellent experience. You need a good codec set, a good media stack that can adaptively switch codecs and manage buffers  and a good ‘point-of-presence’ network for the most part.

I am just going to restrict this post to an overview of the flow.

Enjoy:

click on each image for a larger size (if they are small)

This is a facetime all flow – good, plain, SIP (they use MESSAGE for some proprietary data exchange during the call)

rest is perfect sip.

The protocols are here to see (besides SIP)

Ah here is their 200OK for INVITE

A quick look at their RTP stream:

Good Job Apple. Thanks for putting in an excellent quality, standards compliant SIP client embedded into your dialer experience.

Grab the IMS UA code from the HSC Open Source webpage here

(look for IMS UA for Google Android Entry)

The test bed looked like this:

Caveats

• IMS Sipdroid was tested with a SIPp script for PRACK functionality. Other available IMS clients did not support PRACK functionality.

• PRACK support is implemented for the client side. When the client initiates a call and receives 1xx response with Require: 100rel as header, it will respondwith PRACK. However in case client receives INVITE with Supported:100rel header, it will respond with 180 Ringing without Require: 100relheader.

Usual Disclaimers:

• If something is broken, fix it
• We will try to answer your queries if you are facing problems, but there is no guarantee of timeframes (really, we’d just prefer you debug it yourselves)
• Please don’t ask us for timeframes on what is coming next. It may be tomorrow, it may be next year.

MSRP for Android and RTP stack for Android. And yeah, these releases don’t have additional documentation answering stuff like ‘how do I install’. We hope you know that already :-) And if you need docs on how to use the stacks, please refer to the original open source efforts.

Get them here.

Hi folks, as promised, HSC released the ported SIP UA including the stack. This is an update to our previous ‘stack only’ release. We have overwritten the old release with this new one in our download area. You can get it from here (look for the post titled “SIP UA + Stack for Google Android).

This release includes a short illustrative manual on how to use the SIP client and some things you need to do to configure it (mostly because of the limitation of the android emulator and what it does(not) support as of the current date). We have also included a ported RTP stack with this release. Theoretically, this release is all set for a signalling + media use case. I say theoretically, because it seems the android emulator does not support audio capture, so everytime one tries to start a media conversation, the emulator crashes. Folks @ the android-dev group have confirmed this is currently not supported in the emulator (but works on their actual phone).

So naturally, we had no way of testing it, and therefore, the code is compiled, ready to test, and commented out :-)

Have fun. Some of you have asked us whether we plan to continue working on this. Sure – but no commitments. Some ‘IMS zealots’ in my company have started a side project to now make this UE IMS compliant (we do a lot of work on the IMS side, so a android IMS UE will be neat). When that happens, you will know. Naturally, it will be honoring the GPL way of life, since we will do on top of MJSIP.

Enjoy.

PS: If you intend to report this in other blogs/posts, I welcome you to do so. My request is to credit Hughes Systique Corporation with this port and not me as an individual (that is only fair, because there are others working on the port who are part of the company. Thanks)

Update: Apr 29 2008: UA+Stack code released here

Hey folks, HSC released the android ported mjsip stack source code.

Grab it from here. Look for the attachment to the post titled “Ported GNU Mjsip stack for Google Android”

If you see a “PDF” icon under that post, don’t worry, its actually a ZIP with source in it. Our website folks will fix that annoyance soon.

Note that this is a “developer’s release”. It will allow you to start building applications using the mjsip stack on android. It’s not an “automatic solution for long distance calling” (as some blogs reported it) – it is a tool for developers within you to make such applications.

We also have a working SIP UA we ported on top of this stack (which we called ‘SIPDroid’ – no points for being imaginative here). We are yet to release that port – will do so in a short while.

Note that this is just a SIP stack. There is no RTP included. We did some initial experiments with porting open source RTP stacks – seems very simple. We may just do it later, or you do it and add to this effort :-)