>>> On 3/31/2009 at 10:06 PM, in message <[email protected]>, >>> Cuneyt M <[email protected]> wrote: > On the HD note, everyone seems to be promoting HD voice from polycom to > audiocodes etc. > > What sort of bandwidth is that codec using between two HD enabled polycoms? > > I'm still at the Polycom 330 g711u/a level - polycom phones and > audiocodes gateways. > > The way you explain the tool sounds like we are missing in action. > > And the most important question is how much you paid for your first > house Tony? You got me all curious. > > all the best! > > Tony Graziano wrote: > > You are right. I overcompensated just remembering that I saw 80k being used > for one call. Not all calls will achieve that, but after you add overhead, > it's a "safe number". > > > > In my case I was watching this at my POP where we have a system installed > that can look at the SIP call setup, sees the caller/callee, and can also > show me the RTP separate from the call setup and registration. The remote > user is also on the network, but the call traversed my sipx system so I could > see it. It's pretty cool. It better be, it cost more than my first house. > > > > There is overhead, but the realistic approach I saw was that it did indeed > take 80k, backed up by looking at stats at a dedicated router at the other > end. So since it went to the other end, it was "used", there was no video, > just voice from a Polycom (but in my case it was a HD codec, which consumes a > tad more). > > > > > >>>> "Todd Hodgen" <[email protected]> 03/31/09 8:40 PM >>> > >>>> > > May be off topic, but related to this particular thread discussing > > bandwidth. When one is looking at Bandwidth requirements for lines, it's > > important to discuss where those bandwidth requirements are. For example, > > when you are talking about on an Ethernet wire, 800K for 10 trunks of G711 > > is reasonable. However, that translates to a lower number across a > carriers > > T-1 using HDLC or PPP format, as you lose the Ethernet overhead. There is > > an older discussion, but good one that covers this subject located here - > > http://www.tamos.net/~rhay/overhead/ip-packet-overhead.htm > > > > There is also a good online calculator for determining your bandwidth needs > > on www.erlang.com At the top there is a link to their free calculators > for > > call centers, trunking, etc. > > > > > > > > -----Original Message----- > > From: [email protected] > > [mailto:[email protected]] On Behalf Of Tony Graziano > > Sent: Tuesday, March 31, 2009 3:40 PM > > To: [email protected] > > Cc: [email protected] > > Subject: Re: [sipx-users] Bandwidth.com as SIP-TRUNK for SipX 3.10.3 > > > > Hi, > > > > Hope things are going well for you. My comments are below. > > > > > >>>> Cuneyt M <[email protected]> 03/31/09 5:35 PM >>> > >>>> > > Hi Tony, > > > > Good to hear from you. > > > > I didnt know only 3.11.x and 4 will have native SIP trunking, Do you > > know when is the realistic release date for 4? > > > > **No. I am checking the tracker to see what kind of issues are left, but > > there is also a lot of activity. I would hope sometime in April, but I > > reserve the right to re-guess that too. > > > > So for that kind of volume you need a minimum of 15 trunks and a 20-pack > > of DID numbers. > > > > 20-pack DID numbers because they come in 20? in your example we'll be > > using 15 trunks=lines where each will have a DID. What would be the use > > case for the additional 5 DIDs? > > > > **20 pack of DID's because it's cheaper than 15 when you buy in bulk. If > all > > of your agents are on the phone at once then you will need at least 15 > > trunks. The question comes when you get a busy signal for incoming calls > > because you don;t have enough trunks. Bandwidth.com gives you the ability > to > > add trunks later, but their paperwork process takes about a week to go > > through provisioning. You can always add lines, but you'll want to try to > be > > accurate to start with. > > > > Remember to spare some bandwidth for this, *on tten simultaneous voice > > conversations 9where audio is playing of any sort, not ringing, like > leaving > > or listening to voicemail, AA or speaking interactively with another > person, > > then conservatively you need to budget roughly 800k for bandwidth in > > upload/download, * > > > > **We tracked SIP calls here from a Polycom phone on our network through a > > DPI system. It showed us marvelous things, and we watched in amazement as > > some calls tipped 80k of bandwidth as I recall. When we ran 5 calls > > simulaneously through bandwidth.com, we saw somewhere around 400+k of > > bandwidth being used. So in real case 10 calls needs 800k, 15 needs 1200k > in > > both directions. > > > > I didnt quite get your point here. Can you please re-phrase it within > > actual use-case scenario. You clearly have experience on getting this up > > and running and dont wanna miss what you are saying in the > > red-highlighted sentence. > > > > Regariding blue-highlited sentence, budgeting 800k bandwidth for both > > upload and download is calculated on 15 concurrent user basis? i.e. How > > do you calculate the bandwidth requirement as 800k? Is this an > > estimation based on G711u/a codec (if i do recall correctly g711 > > required 64Kb per line but i could be mistaken) ? > > If the calling end is Philippines and receiving end is > > US(Bandwidth.com), taking the distance and possible-hops between two > > locations which codec usage will be logical? > > And how should I calculate the bandwidth requirement accordingly? > > > > > > Bandwidth.com offered a 30 day trial (not sure yet if its free or with a > > fee) and i was thinking i can play with it in SipX 3.10.3 by adding > > under Gateways. > > > > **It wont work with a sip aware firewall or trunking appliance, free or > not. > > > > I don't have any test system as all live systems are running 3.10.3 at > > the moment and i saw quite a number of issues reported for the 3.11.x > > which scared me to update any of the live systems. As we are on tight > > budget, is there any other way to run Bandwidth.com SIP Trunking without > > Ingate - like an opensource software that runs on sipx or other system? > > I heard about OpenSBC, SIPproxyD but didnt have the chance to look at > > them and what problems they solve. > > Does any of these or other open source software are able to make SipX > > 3.10.3 SIP Trunking work, without Ingate sort of appliance? > > > > **I don't think siproxd is worth any effort. OpenSBC shows some promise but > > it seems a lot to do and is somewhat finicky. I tried twice and gave up > > citing not documented enough yet and the forum was not very helpful unless > I > > wanted to buy a support agreement, at which point I went and got an Ingate, > > learned that, and then cycled through a couple of problematic ITSP's before > > coming to Bandwidth.com. > > > > If it were me, and it's not, Ingate would be a sure bet and would not > > require any changes inside your network and would seamlessly get 3.10.3 > > working with bandwidth.com. If you need to wait for 4.0, I suggest a test > > system gets setup so you can work your way into it, as sipXconfig is much > > changed and there seems to be a bit of a learning curve for me while I > > understand the nuances of sipXbridge and sipxrelay. > > > > Thank you for your prompt reply and very informative e-mail. > > all the best! > > > > Tony Graziano wrote: > > > >>>>> On 3/31/2009 at 4:38 PM, in message <[email protected]>, > >>>>> > > Cuneyt M > > > >>>>> > >>>>> > >> <[email protected]> wrote: > >> > >> > >>> Hello everyone, > >>> > >>> We have existing setup of SipX 3.10.3 with AudioCodes gateways. > >>> > >>> However, we will be testing/integrating Bandwidth.com US Sip-Trunk (with > >>> DID) with SipX. I understand that we can add a Sip Trunk just like > >>> another gateway in Sipx(?) > >>> > >>> > >>> > >> In 4.0 the sipx system is able to communicate with ITSP's, there is > >> > > already a preconfigured template for Bandwidth.com in the current 3.11x > > development version. In 3.10.x you will still need a sip trunking device, > > like an Ingate. > > > >> > >> > >>> The main use-case would be Philippines satellite-office with local SipX > >>> server routing +1 numbers via the Bandwidth.com's SIP Trunk. > >>> I assume we'll need one Sip Trunk per extension (if we are to assume > >>> each extension will be sales guys calling their leads in US) - correct > >>> me if im wrong. > >>> > >>> > >>> > >> Think of "trunks" as "lines". These are call paths. If you have a two-way > >> > > trunk (inbound and outbound calling), then you can either receive or place > a > > call on the trunk (line). If you have 20 people, and 5 are making calls and > > 5 are on calls that came in and 5 more are in the process of ringing, > that's > > "15" simultaneous call paths. So for that kind of volume you need a minimum > > of 15 trunks and a 20-pack of DID numbers. Remember to spare some bandwidth > > for this, on tten simultaneous voice conversations 9where audio is playing > > of any sort, not ringing, like leaving or listening to voicemail, AA or > > speaking interactively with another person, then conservatively you need to > > budget roughly 800k for bandwidth in upload/download, and if you are > sending > > it through a firewall you might look at prioritizing your traffic based on > > destination and port. > > > >> > >> > >>> We have no prior dealing with ITSP or SIP trunking as we used AudioCodes > >>> GWs for local or close-proximity termination. > >>> > >>> Can you please share your thoughts and experience in setting up a Dial > >>> Plan in SipX 3.10.3 where it'll route the call to the registered > >>> ITSP(under Gateways) for numbers that start with +1 (US numbers). > >>> > >>> > >>> > >> I would think that the number would choose the gateway so to speak, seems > >> > > straightforward. > > > >> > >> > >>> Also if you can share your experience on US-based Sip Trunk > >>> providers(Bandwidth.com and others) for service reliability, features > >>> and integration with SipX 3.10.3 i would very much appreciate. > >>> > >>> > >>> > >> We use them now with an Ingate, it's been very stable and pleasant, and > >> > > the config for the Ingate is one of the least complicated I've seen. I do > > think you need a static IP address for bandwidth.com, and it makes > > registration issues a "non-issue". > > > >> > >> > >>> Thanks in advance. > >>> all the best! > >>> _______________________________________________ > >>> sipx-users mailing list > >>> [email protected] > >>> List Archive: http://list.sipfoundry.org/archive/sipx-users > >>> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > >>> > >>> > >>> > >> > >> > >> > >> > >> > > > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > > > > > > > > Trying to attach a screenshot of what I normally see with SIP calls in the way of bandwidth. It's only a 14k attachment, but this is a normal g711ulaw codec using 83.6k in each direction, and i consider that normal.
<<attachment: bandwidth-usage-sip-call.PNG>>
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