Help on RTP

Posted on 2006-05-25
Last Modified: 2008-03-03

Please send me some tutorial on RTP.

Question by:arijit_rebaca

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    LVL 4

    Accepted Solution


    The RFC has got it all, but if just want the important stuff, check out and white you are there, check as well.

    As for a TUTORIAL, nothing beats setting up a couple of ip phones (like the free SJ Phone from and getting them to call each other while capturing the traffic using the free ethereal analyser ( which has some great utilities to help analyse traffic flows, jitter etc.

    As for a two minute tutorial, here are some basic points:

    * RTP packets can carry any kind of encoded voice or video.  As far as voice goes, it is usually 20ms voice per packet.  If this is encoded using G.729 the payload is 20 bytes.  Add to this 12 bytes of RTP header, then 8 bytes of UDP header, plus 20 bytes of IP header and you'll see that it takes about 24kb/s to carry a voice stream (plus layer 2 headers).  On the other hand, if the voice is encded using G.711 (just like the PSTN today), your voice stream will consist of the same header overhead, plus 160 bytes of data per 20ms, or 80kb/s (plus layer 2 headers)

    * Each RTP packet carries a media descriptor, so I can tell whether the packet is carrying G.711, G.729 or other encoding.  The actual decision as to what encoding to use is negotaited during call setup by whatever siganllilng protocol being used (typically SIP or H.323).

    * Also negitiated during call setup are the UDP port numbers that will be used during the conversation.  These can be any port number >1024, so it is impossible to detect RTP traffic at a firewall by port number alone.

    *  RTP is connectionless, so it does not resend lost data.  However, since each packet has a sequence number that increaes by 1 for each packet, the receiving station can tell if any data was lost and compensate for the loss.

    Let me know if you need more detail.


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