iConverged

(July 13: sorry for the downtime, looks like my bandwidth limits were exceeded. Upgraded my hosting package – fixed)

(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.

Hi folks, as promised (over and over earlier we finally have an initial version of the IMS UA ready for download. The IMS system we used was Open IMS

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:

Read the rest of this entry »

Some more goodies from HSC.

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.

Hola,

continuing the “season of giving” for Google’s Android, HSC has also ported the popular java stun stack, JSTUN for the android platform. Download it here. 

Update: Nov-20-2008: Updated SDK + UA for 1.0 SDK released here. Also released jSTUN port for android 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).

Read the rest of this entry »

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

Final update: For folks who are still reaching this post via searches, the [SIPDroid] project took our initial work and have extended it for a full featured application. Instead of downloading our stuff, check them out – their work will be more recent. We (HSC) are not updating our files.

Update: April 2 2008: Source code of stack released here

Update: Mar 17 2008

See here for a screencast

Update: Mar 12 2008

Some other sites linking here seem to be reporting this news with their own verbiage. Lets be specific on what we are doing:

• We have taken the GPL’d mjsip SIP stack and our objective is to make it work on android (this is mostly a porting activity). We are not writing our own stack.
• We are not doing any optimizations, etc. The scope is exactly what I wrote – make mjsip work on android, so developers can use the mjsip APIs to build apps in android.
• It is a very straighforward ‘take from community (GPL/mjsip), give back to community (post ported code back to GPL/mjsip)’ activity.
• We are targetting to release the port in a week or so… (as-is – as I mentioned the objective is straighforward -a port of mjsip – if it has bugs when you do funky stuff , fix it yourself when we release it )

We (my company) had started an internal project to get SIP working on Android and some smart folks belted out some nifty code to get SIP working on android. We hope to release the ported SIP stack on android pretty soon. We used the GPL’d mjSIP stack and will be releasing the modifications as per GPL, obviously – so other developers have a good SIP API to build apps. This should happen pretty soon. Some time ago, we did a rss-to-presence implementation stub (concept here) over Google Mashup editor. I look forward to seeing the “google-phone” talk SIP to the “google-server” and other cool stuff.

Stay tuned!

Update: I have now created a sidebar widget titled “Tech Blogs I read” instead of updating this post. Please continue sending me quality tech blogs (with more focus on concepts and less on marketing) as you come across them and refer to the sidebar for updates and not this post – thx.

Total number of ‘market reporting’ VoIP blogs = k+1

where k=number of times you can blink in a day.

However, there are very few blogs that talk about more technology & architecture details and less market details for all things VoIP, SIP, IMS, web 2.0 (with focus on telecom).

So here is a call to unite!

We need a list of what I call ‘geek-talk’ – those that provide more technical insight into how things are.

Here is my list so far. Please update me /comment here with more tech-blogs and I will update this list

Many of these blogs are a mix of techno-marketing, but are written by people who are neck-deep in actually developing/architecting many of the talked about solutions themselves, and hence offer a more detailed insight.

last updated: Apr-30-2008

1. TurnGeek – focus on P2P, SBC, etc.
2. IMS Lantern – IMS architecture related
3. Voice of VoIPSA – VoIP Security
4. IMS Quality – Testing and Monitoring with focus on IMS
5. VoIP Survivor – general voip, with significant focus on their company products
6. TelCAB – IMS B/OSS
7. iConverged - this blog

I spoke at the Internet Telephony conference in Los Angeles last week on IMS, hWiMAX, how they all work together (or not) and what it really means to applications. You can see a copy here