Jabber, Jitsi, Nimbuzz and XMPP

Event: April 29th, 2011

From Wikipedia:

Extensible Messaging and Presence Protocol (XMPP) is an open-standard communications protocol for message-oriented middleware based on XML (Extensible Markup Language). The protocol was originally named Jabber, and was developed by the Jabber open-source community in 1999 for, originally, near-real-time, extensible instant messaging (IM), presence information, and contact list maintenance. Designed to be extensible, the protocol today also finds application in VoIP and file transfer signaling.

Emil Ivov of Jitsi.org (formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz) join us for a day of Jabber. Here’s a rough outline of what we’ll be talking about.

* What is XMPP/Jabber
* How does one do telephony with XMPP
* How does XMPP/Jingle compare to SIP and (why) is it better.
* Who supports it
* Facebook and their XMPP gateway
* Google Talk
* Nimbuzz – one of the biggest VoIP providers using XMPP as their primary protocol
* NAT traversal
* How does one do it with XMPP
* Again, how is this part different from what we have with SIP
* Media relaying with TURN and Jingle Nodes

And so much more… two hours of it!


Anonymous May 1, 2011 at 11:01 pm

Wow! I just listened to this podcast. I have no affiliation with XMPP and, other than an experimental Jabber IM rollout, no experience with it either.

I was keen to hear what these guys, and XMPP, had to offer. I remain skeptical that this isn’t just a reinvention of the SIP wheel but, I think you guys beat up the guests pretty badly. Please bring them back, if they’ll come, and let them state their case without counter points. At least until the “after hours” free for all.

Also, BOO! to the atrocious audio quality during the call. It’s positively inexcusable that such technically proficient participants of the VUC send such craptastic audio. Yet every week, there is at least on warbling, echoing popping, packet losing, too low, too hot… We really need an audio minimum requirement to participate in the call. It doesn’t have to be g.722 but, listening to people echoing over their mobile while in the toilet has really got to stop.

Have some pride in your vocation/avocation.

Zeeek/Randulo May 2, 2011 at 6:22 am

On audio quality: it’s the old son of a shoemaker whose father has no time to make him shoes. I agree, I think it’s something every caller needs to try to improve for “VoIP Pride”. We could maybe have a virtual 3D parade, too. But seriously, here are but a few very simple rules.

1) Use a headset or a microphone/headphones, always. Not using them introduces echo. Avoid Bluetooth, unless you know your sound is good.
2) Find a way to test your volume level and your sound before speaking live. You’ll be glad you did.
3) Make sure the right microphone is selected. All platforms have this issue.

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