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SIP bandwidth?

Viewing 5 posts - 1 through 5 (of 5 total)
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  • #48857
    Marcin
    Guest

    Hello Folks,

    I am looking for information how to calculate bandwidth consumption used by SIP communication during avarage VoIP conversation. Stress – SIP protocol – no RTP, only signaling. I am digging through Internet, but can’t find anything useful.
    I can understand, that for one line this traffic can be skipped during bandwidth calculation, but having concentration like 100k of subscribers it starts to be an interesting issue.
    Also useful would be information how to tie BHCA values with SIP activity. Could anybody shed some light on that?

    Regards,
    Marcin.

    #48858
    Tom H
    Guest

    The answer is simple – zero. There is no significant SIP communications during a call. SIP is only used for call signalling purposes such as setting up a call, clearing it and a few mid-call features like DTMF transmission.

    The mid-call bandwidth requirement is for RTP (over UDP, over IP).

    #48859
    Marcin
    Guest

    Tom,

    I do not think you got my point. It is quite clear, that during conversation it is close to zero. Using G711 even DTMF may go inband. However it is not zero in given period of time. Read carefully – having 100k subscribers there is a real bandwidth consumption on the line during call setup/clearing, because it is not separated from RTP. So, some of the calls were set up, some are just tearing down and some other are in the process of setting up. In total it generates traffic on the line. In such a case it is not zero if you look carefully at the scale of the traffic. Even if it is 1, or 2 percent it is still some bandwidth consumption. And that’s what I am looking for.
    If you have to calculate system capacity having BHCA there is for sure relation with SIP messages exchange. Because BHCA gives you information on how many people TRIED to set up the call during one hour, and not how many succeeded. You may try to call particular person 5-10 times during busy hour and have no conversation at all, because busy signal, or he/she is out of the office, or simply cannot pick up the call. Comparing to SIP network – you will generate fair amount of signaling traffic and no RTP.
    So, thank you for response, but answer is not that simple for sure 🙂

    Regards,
    M.

    #48860
    Tom H
    Guest

    Yes, I suppose you are right. At some point, it becomes significant.

    I am not aware of any studies on this matter. I suspect that the main focus on SIP signalling has been on its impact on call handling capabilities of equipment (calls handled per second etc.) rather than on its bandwidth consumption.

    #48861
    Marcelo M.
    Guest

    Marcin, in my configuration the Softswitch just work as a MGC, so the RTP go directly from one endpoint to the other without pass through the SSw.
    So is another example where the SIP signaling is bw is relevant, on this case almost 100% of my bandwidth.
    Now I need to connect via internet to another SSw and have your same question.
    Did you obtain any aproach

Viewing 5 posts - 1 through 5 (of 5 total)
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