1. 21 May, 2020 2 commits
  2. 20 May, 2020 2 commits
  3. 15 May, 2020 1 commit
  4. 14 May, 2020 1 commit
  5. 13 May, 2020 1 commit
  6. 12 May, 2020 1 commit
  7. 11 May, 2020 1 commit
  8. 07 May, 2020 2 commits
  9. 06 May, 2020 2 commits
    • Adrien Béraud's avatar
      videomanager: catch exceptions · eb571498
      Adrien Béraud authored
      Change-Id: Ic4a90183ae7589d4ddf7dd196e54cd6e8cba02f3
    • Sébastien Blin's avatar
      pjproject: avoid to fail the negotiation when TURN needs a retry · d10e7a1e
      Sébastien Blin authored
      This aims to fix the some negotiation failure when:
      UserA (TURN ipv4) calls UserB (ipv4 only)
      In fact, the first user will take some time to know the userB
      addresses to allow on the TURN. So, during this time, the TURN will
      give some EPIPE errors. EPIPE is code 32 defined by 120032 in pjsip
      So previous code was a mistake
      Change-Id: If38c5d36dc38bfe89e3b2ccfbf1b64c934639adb
  10. 01 May, 2020 1 commit
    • Sébastien Blin's avatar
      pjproject: fix tls packet reconstruction · fe198fbe
      Sébastien Blin authored
      In the case whare two packets like this were received:
      | data (X bytes) | header (2b) | data (1 byte) |
      | data (X bytes) |
      The RTP loop used the last byte of p1 and the first
      byte of P2 as the header. Leading to malformed packets
      Change-Id: Ie714fcf0bab1e372f7433342ed60ed4e6d20aff3
  11. 30 Apr, 2020 2 commits
  12. 29 Apr, 2020 1 commit
    • Kateryna Kostiuk's avatar
      call: avoid deadlock on hangup · 7d8e4323
      Kateryna Kostiuk authored
      This patch move checkAudio() out of call mutex. Because it could cause
      deadlock if subcallStateChanged() called at the same time.
      Change-Id: Iee06874fb79c8e0953e061f7a669b3f61f13d8b9
  13. 28 Apr, 2020 4 commits
    • Sébastien Blin's avatar
      contrib: bump opendht · 6f687dfa
      Sébastien Blin authored
      Change-Id: I5c991b37011b4c3d97114c9f2b8f05b742e029a3
    • Ming Rui Zhang's avatar
      manager: reset sipLink before pj_shutdown · 9e0de228
      Ming Rui Zhang authored
      it is to prevent pj_shutdown from doing harmful actions to sipLink destruction
      Change-Id: Icb472b2f3350f39f31370a5dba313abe3c8a7bb1
    • Pierre Lespagnol's avatar
      srtp: fix decrypt errors · 52cea6a2
      Pierre Lespagnol authored
      Before this patch when the the sender is restarted, we get the last sequence number from media_encoder that is refreshed on each frame only. But the sequence number continue to be incremented on each RTP packet sent, this result in rollover counter (ROC) desynchronisation in srtp context because the largest sequence number sent is now superior to sequence number that is supposed to occur only when sequence number overflow 65535.
      With the patch when the sender is restarted, we keep the last sequence number from socket_pair (RTP level)
      Change-Id: I531e3d0a073c251c78bbf9f0ffc702aafaf6ccc8
    • Pierre Lespagnol's avatar
      autoadapt: Allow bitrate changes for all API and Codec · 939f6946
      Pierre Lespagnol authored
      Restart encoder when Codec or hardware API don't support dynamic bitrate.
      Change-Id: Ic5fb95d405dc7694125ccb3c80020af39441b2b7
  14. 27 Apr, 2020 1 commit
  15. 24 Apr, 2020 1 commit
  16. 22 Apr, 2020 6 commits
    • Sébastien Blin's avatar
      misc: clean some build warnings · 1bf6c612
      Sébastien Blin authored
      Change-Id: I2dcb7e71738f6f71ebb9db770902d64b24920022
    • Sébastien Blin's avatar
      manager: make manager owns sipvoiplink · 5e086184
      Sébastien Blin authored
      The current design describes SIPVoIPLink as a Singleton. This cause
      some behaviors where the link is destroyed then immediately created
      or vice-versa and its creation/destruction can't really be planned.
      This design was made to allow multiple Managers to exist. However,
      we never do that, so let's keep it simple, there is only one
      Manager and all accounts needs that SIPVoIPLink
      This patch change this behavior and the voip link is now owned
      by the manager.
      Change-Id: I248e41742d342cf452a5503f532fe5ab862166e6
    • Pierre Lespagnol's avatar
      autoadapt: tune parameters for autoadapt algorithm · 0e53344b
      Pierre Lespagnol authored
      - Decrease drops threshold for bitrate diminution (10% to 5%)
      - Space number of decrease from 2 to 1 every 500ms for delays detection
      - Always use bitrate from MediaStream during encoder init
      Change-Id: If3d696b8fc1bc7f6fabe98f70652921d4113b7b7
    • Pierre Lespagnol's avatar
      recorder: use CQ mode for recording · 588a23f7
      Pierre Lespagnol authored
      Use RateMode parameter instead of autoQuality, it makes more sense for the encoder
      Change-Id: Ia5b6acdfb08fb146dec11b75f1d26127a0ed520b
    • Adrien Béraud's avatar
      jamiaccount: don't create connection manager on creation · b9834d78
      Adrien Béraud authored
      Change-Id: I2a1a29a8fb99cf8eea40d35c511dadebd0fb46a2
    • Adrien Béraud's avatar
      account manager: check info · 6bff8561
      Adrien Béraud authored
      Change-Id: Iccae983c6d70ba47e0ec20c83d209c16bbe8b2b0
  17. 20 Apr, 2020 3 commits
  18. 17 Apr, 2020 7 commits
  19. 16 Apr, 2020 1 commit
    • Sébastien Blin's avatar
      call: close audio at the end of a call · 88321904
      Sébastien Blin authored
      hangupCall() remove the calls in a dht pool. This means that the
      operation can take quite some time. So, checkAudio() should be
      done once the call is removed. In the other case, the audio layer
      will remains open forever
      Change-Id: Ia28c99001989215045389e141f694e7732bfcd1d