Sip stack bug


First i will thank Dele for the wonderful work on the Flash phone, for some time I was in contact with Dele regarding the red5 flash development got some useful tips from him. To try boost the development i bring here few problems I have and a possible solution.

my main problem is that I am not java programmer and need a bit help from the community.

unlike many of you I am not using asterisk as my ip pbx, but the equipment I do use is sip complaint more then asterisk.

bug number one found on mjsip was regarding genrating new branch header in sip - RFC 3621 state:

BRANCH in ACK for NON 2xx message should contain the BRANCH of the invite.

there are more problem with wrong branch in mjsip.

reregistering when working against sbc - mjsip send register with expire=3600

in 200ok for the register message sbc ask next registration with expire=30 and mjsip is not sending the next registration as proxy/sbc ask.

there is another great application using open source with mjsip stack when all those problem fixed, please look on : the application name is siptheSkype.

you can find that its easy to fix those bugs in the flash phone sip implementation just by copy few source code lines (i fixed myself the branch problem on the flash phone from there).

problem is after I compile the new sip.jar and make new red5.war I get voice only in one direction.

I would recommend to use the fixed mjsip from SiptheSkype project as it fix and add a lot of sip features in mjsip stack like: call transfer, hold, better nat handling…

the migration should be easy to do to a skilled java programmer.

the result will be a flash phone that is compatible with more sip services (such the one my company offer).

i guess not all the readers of this post might understand what i am telling here about bugs in sip signaling, but if you have some knowledge about sip rfc then just look in ethreal and compare with the standard.