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* RTP: fix few leaks when fail to add streams into a group.Chia-chi Yeh2010-08-191-7/+8
| | | | Change-Id: Iefb3fe219ad48641da37a83c8d14e9ebf1d3086c
* Fix the IN_CALL mode issue.Chung-yih Wang2010-08-162-15/+11
| | | | | | | | If the sip call is on-holding, we should not set the audio to MODE_NORMAL, or it will affect the audio if there is an active pstn call. Change-Id: If1bcba952617bf8427bc9e2d64d483ba1ee37370
* SipAudioCall: perform local ops before network op in endCall()Hung-ying Tyan2010-08-101-1/+4
| | | | Change-Id: I1808f715d56c0979cea7741cb5bdb3831774d3ef
* SIP: clean up unused class and fields.Hung-ying Tyan2010-08-102-82/+0
| | | | Change-Id: I79ed7fb324fea9a52946340055b5ea1d389a926a
* Revert "Move SIP telephony related codes to framework."Chung-yih Wang2010-08-051-2/+1
| | | | This reverts commit b631dcf3eb449ddec756bea330f4e70b996ffb9e.
* Move SIP telephony related codes to framework.Chung-yih Wang2010-08-051-1/+2
| | | | | | + hardcode the sip service for build dependency. Change-Id: Ib0e9717c9b87eb6e06ffa3a7b01ae31184de61bb
* Move the sip related codes to framework.Chung-yih Wang2010-08-0519-0/+3487
Change-Id: Ib81dadc39b73325c8438f078c7251857a83834fe