sipsorcery's blog
Occassional posts about VoIP, SIP, WebRTC and Bitcoin.
Occassional posts about VoIP, SIP, WebRTC and Bitcoin.
There was an announcement a while ago on the xmpp.org site about Google’s adoption of the official Jingle standard. The same announcement was posted by Peter Thatcher a Google engineer to the Jingle mailing list. The announcement got a bit of attention and I originally came across it on Hacker News. After spending a fair bit of time looking into what the upgrade means as far as integrating with Google’s XMPP service the conclusion I’ve come to is not much at least not yet.
The original reason I, a SIP person, started messing around with XMPP and Jingle was to see if there was a better way to integrate with Google Voice. I made a few blog posts at the time and ultimately the investigation didn’t end up yielding any fruit because of a peculiarity in the way the Google XMPP server deals with the RTP media streams which would preclude it working with standard SIP devices (very briefly it requires that STUN packets are exchanged on the RTP sockets before the RTP can flow).
So fast forward 8 months and I finally found some time to delve into the Google XMPP service again to see if there’s any chance of getting SIP interop, or more correctly RTP interop for SIP devices, working. The other thing that spurred me on a bit was Google + and specifically the integration of the Google Talk Gadget into the Google + application. The Google Talk Gadget is not new it’s been in Gmail for quite a while but the integration looks a lot more interesting with Google +, specifically being able to call from a SIP phone into a Google + Hangout (a web based voice/video conference call) would be cool.
As alluded to above the results of my latest little adventure into Google XMPP land haven’t been very fruitful. Yes Google now appear to be sort of using Jingle for their call set up but the media side of things doesn’t appear to have changed at all and in fact the original email from Peter Thatcher did state the work on XEP-0176: Jingle ICE-UDP Transport Method, which is more down to the nuts and bolts of the RTP side of things, is ongoing. In the meantime the XMPP call set up mechanisms used by Google are a combination of Jingle and Gingle (Google’s original implementation of a Jingle like protocol) and crucially the media layer is still Gingle.
My hope is that when Google get the Jingle ICE-UDP transport implemented they will also implement the XEP-0177: Jingle Raw UDP Transport Method which should allow RTP level interop with SIP devices which will in turn allow SIP Proxy services like SIP Sorcery to make and receive calls with Google’s XMPP network. Some SIP servers that proxy media such as Asterisk and FreeSWITCH are already integrating with Google’s XMPP network by implementing the Gingle/XEP-0176 STUN connectivity check mechanisms on the RTP sockets. In SIP Sorcery’s case and also for other non-media proxying servers such as Kamailio the integration is not possible because it then needs the end SIP devices to do the STUN connectivity checks and until ICE support becomes widespread they wont be able to.
Unfortunately this is another example of the really painful issues caused by NAT. If NAT had never been invented and the World had been forced to upgrade to IPv6 the internet would be a much better place :). In the meantime communications protocols like SIP and XMPP end up with as many addendums and extensions to deal with NAT as they do to deal with their core functions!
