Unexpected disconnecting of flash phone used with rtmplite

[GoogleCodeExporter]

Hi to developers of rtmplite!

We have many disconnects using flashvideoio phone with siprtmp server...
Every second call gets disconnected after 15-20 seconds of SIP conversation.

We tried to use X-Lite SIP client instead of flashvideoio + siprtmp.py, with 
same SIP provider, without any disconnections.

We use last, 2.6 version of VideoIO.swf, with r83 version of siprtmp.py, Flash 
Player 10.3

I've attached a typical log with disconnection, please search it for 
"!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!" string - there disconnection (sip unregister 
happens).

Original issue reported on code.google.com by eag...@gmail.com on 8 Sep 2011 at 2:08

Attachments:

• siprtmp_disconnection.log

[GoogleCodeExporter]

Also, if you will require a tcpdump log to find a problem, we can provide it.

Original comment by eag...@gmail.com on 8 Sep 2011 at 2:10

[GoogleCodeExporter]

Hi, 

The rtmp-register after call is initiated is definitely suspicious. The 
register is sent when siprtmp receives a new NetConnection.connect from the 
Flash application. After that it is sent for refreshing the registration every 
3 min by default.

Are there multiple NetConnection.connect (or multiple VideoIO) connecting to 
siprtmp using the same SIP account? The current siprtmp code is written to 
disconnect all other connections when a new connection for same SIP account is 
received. To solve this you can follow the thread 
http://groups.google.com/group/myprojectguide/browse_thread/thread/68dcc4b85e9fb
426/7d3201b86d4b2b5a?hl=en_US&lnk=gst&q=multiple#7d3201b86d4b2b5a
(or search for 'multiple registrations' in that group messages).

Try increasing the registration refresh interval from default 180 (3 min) to 
higher 3600 by changing user.bind(...) to user.bind(..., interval=3600) in 
siprtmp.py. I don't think that the register I see is due to refresh but at 
least it will clear the doubt.

There might have been some incompatible changes between VideoIO and rtmplite, 
so I would recommend using the latest from SVN for rtmplite/p2p-sip and using 
latest of VideoIO.

Finally, if you can send me the sources of the client side HTML/javascript, I 
can test using my set up. If you are worried about confidentiality of your 
sources, feel free to send to my email (kundan10@gmail.com) instead of mailing 
list or attachment here.

Original comment by kundan10 on 8 Sep 2011 at 5:42

• Changed state: Accepted

[GoogleCodeExporter]

Marking as Invalid as this is taken care of...

Original comment by kundan10 on 20 Sep 2011 at 5:40

• Changed state: Invalid