Yes, but, latency is always going to be an issue for this type of user... You need a plan:
Have the remote user run a voip speed test like this from their PC BEFORE they try to use a softphone: http://www.voipreview.org/voipspeedtester.aspx <http://www.voipreview.org/voipspeedtester.aspx>Wait for the report. After the test completes it spits out some basic characteristics of your connection: Speed test statistics --------------------- Download speed: 1190784 bps Upload speed: 5090384 bps Download quality of service: 46 % Upload quality of service: 98 % Download test type: socket Upload test type: socket Maximum TCP delay: 141 ms Average download pause: 14 ms Minimum round trip time to server: 94 ms Average round trip time to server: 94 ms Estimated download bandwidth: 28000000bps Route concurrency: 23.51392 Download TCP forced idle: 80 % Maximum route speed: 5577440bps VoIP test statistics -------------------- Jitter: you --> server: 8.5 ms Jitter: server --> you: 9.2 ms Packet loss: you --> server: 0.0 % Packet loss: server --> you: 0.2 % Packet discards: 0.0 % Packets out of order: 0.0 % Estimated MOS score: 3.7 The second graph is quite compelling, showing you where you might have broken sound or an unuesable connection. You need a plan... a secondary way to connect to a pstn connection that is "closer" in peering than your voip server in the US... I think Skype and Google dial (from gmail, or even google voice) have already been mentioned. I don't think the fact that they are there temporary or not matters. Anytime you travel, this can happen (overseas or not). I have google voice on my cell phone, laptop and a completely separate gmail account with the ability to dial US numbers for free (outbound only) too... preparation is key. You never know who is blocking what when you travel some places, or how the ITSP in some countries are required to block SIP, or treat SIP on connections where they are not the itsp and de-prioritize the traffic to sip connections that they do not sell to you... On Tue, Oct 5, 2010 at 11:57 AM, Stiles Watson <[email protected]>wrote: > Sorry, I should have mentioned that the VIP is traveling overseas, not > permanently stationed there. > > codec: u-law > > I'm not sure what his ISP is as he is using the hotel's Internet > connection. > > Stiles > > Michael Broome wrote: > > What codec are you using and whats the ISP provider > > M Broome > > *CEO* > > SATEL, inc > > (800) 591-7033 > > > > > > > > > > > > > > On Tue, Oct 5, 2010 at 10:36 AM, Stiles Watson <[email protected] > > <mailto:[email protected]>> wrote: > > > > I've got a VIP overseas using X-lite to place calls over the public > > Internet back to the States via our sipX. Calls go through fine, > > but he > > says that the some callees tell him that there is a bad echo (all > hear > > themselves to some degree). All outbound calls go though our ITSP > > (flowroute). All callees are on PSTN or Cell. > > > > Is there any way to reduce this? > > > > Stiles > > _______________________________________________ > > sipx-users mailing list > > [email protected] <mailto: > [email protected]> > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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