jamie prince
15e266cb43
- AudioRouteManager: route to onboard speaker when no BT headset is connected, BT SCO when one is; set MODE_IN_COMMUNICATION + focus. - aaudio: build with AAUDIO_PERFORMANCE_MODE_NONE so Android routing (speakerphone) is honoured instead of being bypassed by low-latency. - DialerScreen: restyled to a stock-dialer layout (large number display, 3x4 keypad with sub-letters, green call FAB, backspace, recent chips). - InCallScreen: added DTMF keypad (sendDigit) for voicemail/IVR. - WearBaresipService: on incoming call, wake screen + vibrate + bring activity to foreground so Answer/Decline is visible. - Fixed dial normalization (bare extension -> full SIP URI with domain) and the uap parse in uaEvent so defaultUap is set and calls connect. - config.static: load g711/opus codecs to resolve 488 Not Acceptable Here. - Telecom scaffold (experiment): PhoneAccount + ConnectionService handed incoming calls via addNewIncomingCall to test stock Wear dialer pickup. - Submodule libbaresip-android rebuilt with g711/opus/aaudio changes. Verified: assembleDebug green, installs and registers on Galaxy Watch, incoming call event fires and is handled without crashing.
43 lines
832 B
Plaintext
43 lines
832 B
Plaintext
poll_method epoll
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call_local_timeout 60
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call_max_calls 4
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call_hold_other_calls yes
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filter_registrar udp,tcp,tls,ws,wss
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audio_player aaudio,nil
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audio_source aaudio,nil
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audio_alert aaudio,nil
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audio_level no
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ausrc_format s16
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auplay_format s16
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auenc_format s16
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audec_format s16
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audio_buffer 20-160
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audio_silence -35.0
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audio_telev_pt 101
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audio_jitter_buffer_type adaptive
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audio_jitter_buffer_ms 100-200
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audio_jitter_buffer_size 50
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rtp_stats no
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rtp_timeout 60
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rtp_rxmode thread
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module aaudio.so
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module g711.so
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module opus.so
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module stun.so
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module turn.so
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module ice.so
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module srtp.so
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module dtls_srtp.so
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module gzrtp.so
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module uuid.so
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module_app account.so
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module_app debug_cmd.so
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module_app mwi.so
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opus_samplerate 16000
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opus_stereo no
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opus_sprop_stereo no
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opus_cbr no
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opus_inbandfec yes
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opus_application voip
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dtls_srtp_use_ec prime256v1
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