Some Problems solved..
Hey Again,
The problems we had with gstreamer and using its elements are now solved. We had problems with the voice comming through the RTP protocol. That was actually because that the RTBbin that we used was been obsolete and deprecated. Now we use udpsrc etc. (see the source code), which acutally make the voice go through over RTP!. *GREAT!*
Now we are only missing to incoorporate it into our plugin code. This is not done yet because of some heavy research regarding NAT and getting the public IP address.
As I (zool) spend last week of researching how to bypass the NAT, I have found some mechanisms which I only will mention briefly here. I have also writte a section in the report about the same area:
*STUN
*TURN
*UPnP
*ICE
....
But better yet I found that google (gtalk) and xmpp finally have something in common now. This is pretty new I think:
http://www.xmpp.org/extensions/xep-0215.html
http://code.google.com/apis/talk/jep_extensions/jingleinfo.html
These links show how STUN servers are found via. XMPP. After we have found the server list, we can connect (Via. a STUN client) with one of them. The one thing I dont get is that I have seen many places that states that gtalk is using ICE. I have to investigate further than this. **13/8-2007* The day after (today), I already found out of the context of ICE. ICE is the technique for using STUN for getting the public IP addr. and using TURN for data traversal through nat.
I talked (via. IRC, #pidgin) with a lot of the developers for the pidgin platform, which have connections to the xmpp organization (Sean Egan, utopia etc), and found out that no standard or draft papers are available for conferencing of voice or video. There for now I am involved in making findings of already available software in this area, and find out what kind of demands that can be put. After this I will write a XEP-### for the draft papers.!!! Jubiii, we can be official xmpp developers.. .*GREAT*!...
Well thats all for now!
10-4
/zool (Steffen)
The problems we had with gstreamer and using its elements are now solved. We had problems with the voice comming through the RTP protocol. That was actually because that the RTBbin that we used was been obsolete and deprecated. Now we use udpsrc etc. (see the source code), which acutally make the voice go through over RTP!. *GREAT!*
Now we are only missing to incoorporate it into our plugin code. This is not done yet because of some heavy research regarding NAT and getting the public IP address.
As I (zool) spend last week of researching how to bypass the NAT, I have found some mechanisms which I only will mention briefly here. I have also writte a section in the report about the same area:
*STUN
*TURN
*UPnP
*ICE
....
But better yet I found that google (gtalk) and xmpp finally have something in common now. This is pretty new I think:
http://www.xmpp.org/extensions/xep-0215.html
http://code.google.com/apis/talk/jep_extensions/jingleinfo.html
These links show how STUN servers are found via. XMPP. After we have found the server list, we can connect (Via. a STUN client) with one of them. The one thing I dont get is that I have seen many places that states that gtalk is using ICE. I have to investigate further than this. **13/8-2007* The day after (today), I already found out of the context of ICE. ICE is the technique for using STUN for getting the public IP addr. and using TURN for data traversal through nat.
I talked (via. IRC, #pidgin) with a lot of the developers for the pidgin platform, which have connections to the xmpp organization (Sean Egan, utopia etc), and found out that no standard or draft papers are available for conferencing of voice or video. There for now I am involved in making findings of already available software in this area, and find out what kind of demands that can be put. After this I will write a XEP-### for the draft papers.!!! Jubiii, we can be official xmpp developers.. .*GREAT*!...
Well thats all for now!
10-4
/zool (Steffen)
posted by Unknown at 8:38 PM
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