Modifying T1 timer

Discussion in '3CX Phone System - General' started by CloudSurfer, Dec 17, 2007.

  1. CloudSurfer

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    I live in Samoa in the South Pacific where our link to the world is via satellite. The usual ping times to my VoIP server in Australia is about 830 ms. This creates significant latency and not great voice quality but it is generally acceptable. To improve the quality I have had to change several settings on my Sipura 3000. Most importantly, I must have the T1 timer set at 2 sec or more or I cannot initiate calls. Other changes are to increase the jitter buffer to maximum and set the RTP packet size to 0.080 (default 0.030). I have not downloaded the 23MB of file as yet because it will be useless to me if I cannot adjust these settings in the VoIP setup. None of these is mentioned in the manual.

    Does anyone know if these settings are available in Version 5 or if not, are they are being considered in a future version?

    Ian.
     
  2. BJReplay

    BJReplay New Member

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    Hi Ian, I'm sure there are compensations in Samoa that make up for the lag :)

    There are no user interface settings around these SIP timings in 3CX itself.

    However, all is not lost: 3CX (by default, in V5, as far as I know), is set up to proxy the media stream, but you can override this option. When overridden, the only time the media stream is handled by 3CX is when doing things like putting a caller on hold or a caller going through to voicemail.

    This means that if you don't specify that 3CX handles audio, it lets the SIP end points negotiate - e.g. your SIP phone and your provider.

    This works well.

    For example, I'm currently running 3CX V5 free version (waiting for SLA support before I pony up), and can happily initiate calls using G729a even though this codec isn't supported by the free version - my phone negotiates this with my VoIP Provider. Similarly MyNetPhone likes an RTP packet size of 0.040, and this works without any issues.

    So, if you've got the time and the bandwidth, I'd suggest it might be worth a look.

    Cheers

    BJ
     
  3. CloudSurfer

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    Dear BJ,

    Thanks for the heads up on that. This sounds like an interesting way of managing it. I have an old Gransdstream 286 but that can't initiate calls here because of the latency problem. I was going to set up this and my Sipura 3000 as two extensions as a start and then see whether it all worked well enough to go from there. What I would need if I follow your plan is to get another Sipura/Linksys 3XXX, whatever the number is now, and use that as my second extension. That wouldn't be an all bad idea.

    I did download the free programme and install it only to find that there were significant problems getting it to work with my VoIP provider. When I then deleted the VoIP provider I found the programme kept the information and kept trying to register. Maybe I did something wrong. At present, it is uninstalled and I might wait till the weekend to try again.

    What would be great is if there was an optional page of parameters associated with each VoIP provider that could be altered, as one finds with the Sipura. I am sure in the US that it all works well. I saw someone in one post complaining about a latency of 150ms! I wish! Sadly radio waves travel at a finite speed and geostationary satellites are a long way away. Also, with lots of hops, my transmissions get much more variation in delays. When your latency is short, and you have wide pipes, there is little jitter. Not so when your latency is 830ms.

    Samoa does have its advantages otherwise.

    All the best,
    Ian.
     
  4. SY

    SY Well-Known Member
    3CX Support

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    Do you have any issues with binding to media server? (except using G729)

    Just for information. In "bind to media server"("PBX deliver audio") mode PBX should accept any packet time and should provide packets with time expected(declared) by remote party. Do you have any issues related to ptime?

    Thanks
     

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