1. 11 Feb, 2008 1 commit
    • Emmanuel Milou's avatar
      Codecs no longer loaded at each frame. · 4cab06ab
      Emmanuel Milou authored
      The codecs are loaded at the initialization of the rtp session and destroyed at the end.
      GSM audio quality is far better now. Because the dynamic library had to load at every 20ms the libgsm, it introduced a delay at each frame
      4cab06ab
  2. 08 Feb, 2008 1 commit
  3. 07 Feb, 2008 2 commits
  4. 04 Feb, 2008 2 commits
  5. 28 Jan, 2008 1 commit
  6. 25 Jan, 2008 2 commits
  7. 24 Jan, 2008 1 commit
  8. 21 Jan, 2008 1 commit
  9. 17 Jan, 2008 2 commits
  10. 27 Nov, 2007 1 commit
    • Emmanuel Milou's avatar
      use sampling frequency and frame size from the user config · fa57bac5
      Emmanuel Milou authored
      Sampling rate values are no more hardcoded. The sampling rate of the audio layer
       and the frame size can be set in the user config file. The clock rate of the codec we use
      in the rtp session is set with his actual value, but can be changed by modifying
      the available codec in the user config file (only G711 for now)
      fa57bac5
  11. 21 Sep, 2007 1 commit
  12. 04 Sep, 2007 1 commit
  13. 31 Aug, 2007 1 commit
  14. 23 Aug, 2007 4 commits
  15. 06 Sep, 2006 1 commit
  16. 04 Sep, 2006 1 commit
  17. 30 Aug, 2006 2 commits
  18. 07 Aug, 2006 1 commit
  19. 03 Aug, 2006 1 commit
  20. 02 Aug, 2006 1 commit
  21. 03 May, 2006 1 commit
    • yanmorin's avatar
      · adda9e4b
      yanmorin authored
      Fix onhold/getcallstatus issues
      adda9e4b
  22. 02 Apr, 2006 1 commit
    • yanmorin's avatar
      · f39e33ae
      yanmorin authored
      New account implementation.
      Need to test more and add dynamic handling in Qt Interface.
      SipCall, Call, VoIPLink were rewritten completly
      Configuration now in [SIP0] section of .sflphonedrc
      The debug of sflphoned is very verbose.
      sflphone-cli may not work right now (need to send SIP0 as accountId).
      f39e33ae
  23. 30 Mar, 2006 1 commit
    • yanmorin's avatar
      · d62caf5e
      yanmorin authored
      VoIPLink integration inside account
      getAccount() patch for managerimpl
      Remove old voiplink vector in managerimpl
      d62caf5e
  24. 22 Mar, 2006 1 commit
    • yanmorin's avatar
      · 455a25c3
      yanmorin authored
      New STUN logic to discover true firewall port (NAT)
      Send true IP/port in SDP message
      Remove big ###### debugging message
      New logic to listening on SIP port: 2 tries
      455a25c3
  25. 19 Dec, 2005 1 commit
  26. 18 Dec, 2005 1 commit
  27. 04 Nov, 2005 1 commit
    • yanmorin's avatar
      · b39a205e
      yanmorin authored
      Remove debug message
      Add changes to project files
      Fix a nonat-nonat bug (reinvite)
      b39a205e
  28. 02 Nov, 2005 1 commit
    • yanmorin's avatar
      · 451e8b48
      yanmorin authored
      Fixe issue for current call id switching and for incoming call signal
      451e8b48
  29. 01 Nov, 2005 1 commit
    • yanmorin's avatar
      · 897f34d9
      yanmorin authored
      Only switch call when no one are selected
      897f34d9
  30. 31 Oct, 2005 1 commit
    • yanmorin's avatar
      · 40482ea8
      yanmorin authored
      ReInvite implementation for two no-nat telephone
      40482ea8
  31. 29 Oct, 2005 2 commits
    • yanmorin's avatar
      · afeae31b
      yanmorin authored
      Handling onhold answering call
      afeae31b
    • yanmorin's avatar
      · a701fe3b
      yanmorin authored
      Don't close sflphone when portaudio failed to load
      Remove enable_audio variable (non-protected) in SipCall
      a701fe3b