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Wed 01 of Aug, 2007 [09:12 UTC]

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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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SDP

SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SDP is used from VOIP signalling protocols like SIP, H.323 and some minor VOIP protocols to transfer media setup information about a multi media client from A to B.

  • SDP is used by SAP - the Service Announcement Protocol.
  • SDP is used by SIP

IETF RFCs


Fields

Optional items are marked with a `*'.

Session description

       v=  (protocol version)
       o=  (owner/creator and session identifier).
       s=  (session name)
       i=* (session information)
       u=* (URI of description)
       e=* (email address)
       p=* (phone number)
       c=* (connection information - not required if included in all media)
       b=* (bandwidth information)
       One or more time descriptions (see below)
       z=* (time zone adjustments)
       k=* (encryption key)
       a=* (zero or more session attribute lines)
       Zero or more media descriptions (see below)

Time description

       t=  (time the session is active)
       r=* (zero or more repeat times)

Media description

       m=  (media name and transport address)
       i=* (media title)
       c=* (connection information - optional if included at session-level)
       b=* (bandwidth information)
       k=* (encryption key)
       a=* (zero or more media attribute lines)

RTP Payload Type Numbers

RTP payload type numbers appear in the m= and rtpmap lines of the SDP, but these numbers are not defined in the SDP RFCs, there is more information here: RTP.

See also

  • SIP: The Session Initiation Protocol
  • SAP: The Session Announcement Protocol
  • IETF: The Internet Engineering Task Force
  • IANA: IANA registers SDP parameters

Created by koehler, Last modification by admin on Sun 11 of Mar, 2007 [06:07 UTC]

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