Possible to change codec on-the-fly?

Discussion in 'VOIP' started by Jack L., Feb 23, 2004.

  1. Jack L.

    Jack L. Guest

    Hello group.

    Idea: Is it possible to write a small daemon which changes the codec during
    a VoIP session on-the-fly. This way, while the connection is good the daemon
    chooses the codec with minimum compression and when the connection is bad
    (eg. caused by sudden overload in the routers) the daemon chooses a codec
    with higher compression. It will require to open a channel where both ends
    send data back and forth to determine the traffic along the path. The daemon
    can be used in an existing software such as Linphone or Gnophone.

    While anything is feasible, is it possible for a computer science student to
    finish this application within 3 months?

    --
    Mvh / Best regards,
    Jack, Copenhagen

    The email address is for real. :)
     
    Jack L., Feb 23, 2004
    #1
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  2. "Jack L." <> wrote in
    news:4bw_b.96437$:

    > Idea: Is it possible to write a small daemon which changes the codec
    > during a VoIP session on-the-fly.


    Such functionality is actually part of the RTP specification. If terminals
    (hardware or software) are supporting it is another question.

    It should however be possible to implement RTP proxies for this purpose and
    then route the calls through these.


    Tor
     
    Tor-Einar Jarnbjo, Feb 24, 2004
    #2
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  3. In article <Xns9499449CE2A4TorEinarJarnbjopobox@130.133.1.4>,
    Tor-Einar Jarnbjo <> wrote:
    >"Jack L." <> wrote in
    >news:4bw_b.96437$:
    >
    >> Idea: Is it possible to write a small daemon which changes the codec
    >> during a VoIP session on-the-fly.

    >
    >Such functionality is actually part of the RTP specification. If terminals
    >(hardware or software) are supporting it is another question.
    >
    >It should however be possible to implement RTP proxies for this purpose and
    >then route the calls through these.


    You can also, according to spec, do a re-invite in SIP to change all
    attributes of a session, including transferring calls and this sort of
    stuff.

    Whether the equipment you have will handle it well is another question!
    --
    Brad Templeton's Photography page
    http://www.templetons.com/brad/photo
     
    Brad Templeton, Feb 24, 2004
    #3
  4. Jack L.

    Jack L. Guest

    Tor-Einar Jarnbjo wrote:
    > Such functionality is actually part of the RTP specification. If
    > terminals (hardware or software) are supporting it is another
    > question.
    >
    > It should however be possible to implement RTP proxies for this
    > purpose and then route the calls through these.
    >


    Tor, thanks for your response. Can you give me some keywords, or even better
    direct links, to the functionality in the RTP specification you have
    mentioned? In short, what is the exact term to the functionality?

    --
    Mvh / Best regards,
    Jack, Copenhagen

    The email address is for real. :)
     
    Jack L., Feb 24, 2004
    #4
  5. "Jack L." <> wrote in
    news:2KP_b.96677$:

    > Tor, thanks for your response. Can you give me some keywords, or even
    > better direct links, to the functionality in the RTP specification you
    > have mentioned? In short, what is the exact term to the functionality?


    RTCP (Real Time Control Protocol). It's defined as part of the RTP standard
    and offers you the ability to compute the network latency and to exchange
    connection statistics as packet loss and jitter between client and server.

    ftp://ftp.rfc-editor.org/in-notes/rfc3550.pdf


    Tor
     
    Tor-Einar Jarnbjo, Feb 24, 2004
    #5
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