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 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
