summaryrefslogtreecommitdiffstats
path: root/voip
Commit message (Collapse)AuthorAgeFilesLines
* RTP: Enable GSM-EFR codec.Chia-chi Yeh2010-09-304-3/+110
| | | | Change-Id: I9d84009e4557a0a82c1f9d7d543922741be97c77
* RTP: Revise the workaround of private addresses and fix bugs.Chia-chi Yeh2010-09-301-27/+24
| | | | Change-Id: Ie654b569f47049aa452eca8d3e6d4a98ac18469c
* Merge changes Iae1913fb,I38dbefef into gingerbreadChia-chi Yeh2010-09-285-123/+224
|\ | | | | | | | | | | * changes: RTP: Enable GSM codec. RTP: Refactor out G711 codecs into another file.
| * RTP: Enable GSM codec.Chia-chi Yeh2010-09-294-4/+84
| | | | | | | | Change-Id: Iae1913fb0643f1c66b5d16f24d51924d363e5ef5
| * RTP: Refactor out G711 codecs into another file.Chia-chi Yeh2010-09-293-120/+141
| | | | | | | | Change-Id: I38dbefef2315a28d44683e86a51e69f38e3f20ec
* | Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into ↵Chia-chi Yeh2010-09-281-133/+123
|\ \ | | | | | | | | | gingerbread
| * | RTP: Delay the initialization of AudioTrack and AudioRecord.Chia-chi Yeh2010-09-291-133/+123
| |/ | | | | | | | | | | Related to http://b/3043844. Change-Id: I2c4fd9f64e6eba597d68b2ea1ceeff83103697db
* | SIP: Feedback any provisional responses in addition to RINGHung-ying Tyan2010-09-291-1/+5
| | | | | | | | | | | | | | | | | | The only exception is TRYING. Also remove an unused import in SipSessionGroup. http://b/issue?id=3021865 Change-Id: I160982b0c4b417362f1fb961217db90c3a585ce5
* | SIP: add DisconnectCause.SERVER_ERRORHung-ying Tyan2010-09-281-5/+2
|/ | | | | | | | | and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not from local exceptions. http://b/issue?id=3041332 Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
* Move SipService out of SystemServer to phone process.Hung-ying Tyan2010-09-284-0/+3308
| | | | | | | Companion CL: https://android-git/g/#change,70187 http://b/issue?id=2998069 Change-Id: I90923ac522ef363a4e04292f652d413c5a1526ad
* Merge "SipAudioCall: remove SipManager dependency." into gingerbreadHung-ying Tyan2010-09-272-11/+12
|\
| * SipAudioCall: remove SipManager dependency.Hung-ying Tyan2010-09-242-11/+12
| | | | | | | | Change-Id: I2dc8bf427e52f64529ee0e0261362b975a8917c6
* | SipService: handle cross-domain authentication errorHung-ying Tyan2010-09-271-0/+5
| | | | | | | | | | | | | | | | | | and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK DisconnectCause. http://b/issue?id=3020185 Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
* | Fix the unhold issue especially if one is behind NAT.Chung-yih Wang2010-09-271-2/+27
|/ | | | | | +call startAudio() when call is established. Change-Id: Ib6a1e34017fb83007ce275da1991058e8b803833
* SDP: remove dead code.Chia-chi Yeh2010-09-243-530/+0
| | | | Change-Id: I2a5764a2b9cabc54b0ac18666e494c1cb39c4e9b
* Refactoring SIP classes to get ready for API review.Hung-ying Tyan2010-09-246-1061/+1459
| | | | | | | | | | | | | + replace SipAudioCall and its Listener interfaces with real implementations, + remove SipAudioCallImpl.java, most of it is has become part of SipAudioCall, + add SipSession and its Listener classes to wrap ISipSession and ISipSessionListener, + move SipSessionState to SipSession.State, + make SipManager keep context and remove the context argument from many methods of its, + rename SipManager.getInstance() to newInstance(), + rename constant names for action strings and extra keys to follow conventions, + set thread names for debugging purpose. Change-Id: Ie1790dc0e8f49c06c7fc80d33fec0f673a9c3044
* Fix the build.repo sync2010-09-231-1/+1
| | | | Change-Id: I82210cb2d41f532583f83ea17e6f2d8d49280a30
* SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp.repo sync2010-09-231-196/+185
| | | | Change-Id: I7efff4f29ca84c3e7c17ef066b7186b514a777b2
* SDP: Add a simple class to help manipulate session descriptions.Chia-chi Yeh2010-09-231-0/+612
| | | | Change-Id: I1631ee20e8b4a9ad8e2184356b5d13de66e03db1
* RTP: Add log throttle for "no data".repo sync2010-09-231-1/+5
| | | | Change-Id: I14d9886a40fa780514cbc6c5bac6fb2a670f55f4
* RTP: Update native part to reflect the API change.Chia-chi Yeh2010-09-233-39/+48
| | | | Change-Id: Ic2858920ad77d7312f2429f89ca509a481363431
* RTP: Add two getters to retrieve the current configuration from AudioStream.Chia-chi Yeh2010-09-231-1/+20
| | | | Change-Id: Iff588130653242f6ddd6a6b663df775ecb276768
* RTP: Extend codec capability and update the APIs.Chia-chi Yeh2010-09-234-62/+244
| | | | Change-Id: I37ba9d83c2de3c5dae2bfc1b7513df7f6fee3c5c
* SipPhone: fix missing-call DisconnectCause feedbackHung-ying Tyan2010-09-201-8/+8
| | | | | | | | | also fix delivering bad news before closing a SipAudioCallImpl object so that apps can get the current audio-call object state before it's closed: http://b/issue?id=3009262 Change-Id: I94c19dae8b4f252de869e614ec462b19b4ff2077
* SIP: convert enum to static final int.Hung-ying Tyan2010-09-209-71/+127
| | | | | | Converts SipErrorCode and SipSessionState. Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
* SIP: add config flag for wifi-only configuration.Hung-ying Tyan2010-09-201-0/+8
| | | | | | http://b/issue?id=2994029 Change-Id: I328da9b0f8b70d660dbcefffdac8250341792101
* SipAudioCall: expose startAudio()Hung-ying Tyan2010-09-172-61/+72
| | | | | | so that apps can start audio when time is right. Change-Id: I7ae96689d3a8006b34097533bc2434bc3814b82a
* Add timer to SIP session creation process.Hung-ying Tyan2010-09-174-27/+81
| | | | | | | | | | | | + add timer parameter to ISipSession.make/changeCall(), + add timer paramter to SipAudioCall.make/answer/hold/continueCall()'s, + add timer parameter to SipManager.makeAudioCall(), + modify implementation in SipSessionGroup, SipAudioCallImpl accordingly, + make SipPhone to use it with 8-second timeout. http://b/issue?id=2994748 Change-Id: I661a887e5810087ddc5e2318335e2fa427f80ec6
* Fix links in SIP API javadoc.Hung-ying Tyan2010-09-167-45/+40
| | | | Change-Id: I839280fe18502bb576f6e9c9a7948077c02fa570
* SIP: add PEER_NOT_REACHABLE error feedback.Hung-ying Tyan2010-09-151-0/+3
| | | | | | http://b/issue?id=3002033 Change-Id: Ib64b08919d214acbab89945ac19dc113a68e62ad
* SipService: ignore connect event for non-active networks.Hung-ying Tyan2010-09-151-11/+12
| | | | | | + sanity check and remove redundant code. Change-Id: I4d3e226851ad7fc4d88ddcd0a5c58f7e33b6c14a
* SipAudioCall: use SipErrorCode instead of string in onError()Hung-ying Tyan2010-09-142-24/+52
| | | | | | and fix callback in setListener(). Change-Id: Ic2622df992a2ad45cb1e3f71736f320897ae8fb3
* SIP: remove dependency on javax.sipHung-ying Tyan2010-09-144-12/+16
| | | | | | | and change errorCodeString to errorCode in SipRegistrationListener.onRegistrationFailed(). Change-Id: Id9618f5a4b0effaed04f8b0dc60347499d9e4501
* SipService: deliver connectivity change to all sessions.Hung-ying Tyan2010-09-141-1/+4
| | | | | | | | | + add DATA_CONNECTION_LOST to SipErrorCode + convert it to Connection.DisconnectCause.LOST_SIGNAL in SipPhone http://b/issue?id=2992548 Change-Id: Ie8983c1b81077b21f46304cf60b8e61df1ffd241
* SIP: enhance timeout and registration status feedback.Hung-ying Tyan2010-09-131-1/+7
| | | | | | | http://b/issue?id=2984419 http://b/issue?id=2991065 Change-Id: I2d3b1dd3a70079ff347f7256f4684aea07847f4e
* SIP: remove dependency on javax.sip.SipException.Hung-ying Tyan2010-09-134-7/+40
| | | | Change-Id: I77d289bef1b5e7f1ec0c0408d0bbf96c21085cd7
* SIP: add SipErrorCode for error feedback.Hung-ying Tyan2010-09-106-25/+68
| | | | Change-Id: I8b071d4933479b780a403d0bfa30511f4c23ca8f
* Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbreadChia-chi Yeh2010-09-091-2/+2
|\
| * RTP: prevent buffer overflow in AudioRecord.Chia-chi Yeh2010-09-081-2/+2
| | | | | | | | | | | | | | | | This change simply reduces the receive timeout of DeviceSocket. It works because AudioRecord will block us till there is enough data, which makes AudioSocket overlap AudioRecord. Change-Id: I4700224fb407e148ef359a9d99279e10240128d0
* | SipManager: always return true for SIP API and VOIP support query.Hung-ying Tyan2010-09-031-0/+6
|/ | | | | Change-Id: I397a804e0aa598aee77a8ce28ada1b11e10fbaea http://b/issue?id=2972054
* Merge "SipService: reduce the usage of javax.sdp.*." into gingerbreadChia-chi Yeh2010-09-027-38/+27
|\
| * SipService: reduce the usage of javax.sdp.*.Chia-chi Yeh2010-09-027-38/+27
| | | | | | | | | | | | After this change, SipAudioCallImpl is the only place still using it. Change-Id: I5693bffa54f9e19cbfa70b45dfcf40fba04dedbb
* | SipProfile: remove outgoingCallAllowed flag.Hung-ying Tyan2010-09-021-23/+1
|/ | | | Change-Id: I37a215bafce57adf6911c81fd38db324bac686ec
* Add software features for SIP and VOIPHung-ying Tyan2010-09-021-9/+24
| | | | | | and block SipService creation and SIP API if the feature is not available. Change-Id: Icf780af1ac20dda4d8180cea3e5b20e21a8350bc
* Add Wifi High Perf. mode during a call.Chung-yih Wang2010-08-261-1/+28
| | | | | | | To prevent the wifi from entering low-power mode due to the screen off triggered by the proximity sensor. Change-Id: I490bc594d800bc30c256e52ef3bce08bf86bc7b1
* Merge "Revert "RTP: integrate the echo canceller from speex."" into gingerbreadChia-chi Yeh2010-08-252-18/+6
|\
| * Revert "RTP: integrate the echo canceller from speex."Chia-chi Yeh2010-08-262-18/+6
| | | | | | | | This reverts commit 4ae6ec428f3570b9020b35ada6a62f94af66d888.
* | Add dynamic uid info for tracking the sip service usage.Chung-yih Wang2010-08-261-0/+19
| | | | | | | | Change-Id: Ibc340401b63799326b08aee6eba602a3e753b13f
* | Merge "SipProfile: add isOutgoingCallAllowed() and new builder constructor" ↵Hung-ying Tyan2010-08-241-1/+40
|\ \ | | | | | | | | | into gingerbread
| * | SipProfile: add isOutgoingCallAllowed() and new builder constructorHung-ying Tyan2010-08-241-1/+40
| | | | | | | | | | | | Change-Id: I7ced47079fd2b00c7160b152eb4c1d34399e39dc
generated by cgit v1.1 at 2024-09-10 16:14:40 +0000