Sip Trunking Getting It Right The 1st Time
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Sip Trunking Getting It Right The 1st Time
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Sip Trunking Getting It Right The 1st Time from ITEXPO West 2010
Graham FrancisSeguir
Sip Trunking Getting It Right The 1st Time
- SIP Trunking Getting it Right the First time…! Graham Francis
- SIP is Everywhere! Even here
- ITSPSIPSIP SIP SIP PBX SBC
- Such an old joke now!
- The North American VoIP access and SIP trunking services market User base from 3.8 million users to 46 million users in 2016. Data from Frost & Sullivan
- It’s an Obsession
- 40% – 80% savings
- 23 Lines Reqd
- Lines – You choose DIDs – How many? Disaster Recovery Branch 2 Branch $0? Migrate - T1/E1 if nervous
- Line Side – Trunk Side
- Unified ClientsUnified Clients Unified Server Inc. Registrar and Location services Unified Server Inc. Registrar and Location services SIP IP PhonesSIP IP Phones DirectoryDirectory DNSDNS Messaging ServerMessaging Server GatewayGateway PBXPBX Firewall / NATFirewall / NAT ITSPITSP
- • Voice and Data on Same Network • VLANS Implemented • TOS for Quality across the LAN • Ongoing Monitoring
- • Bandwidth and Codecs • All about ‘The Edge’ • Multiple Sites • Which ITSP? • Stay ‘On-Net’ • SIP Connect
- Bandwidth + Codecs
- Call Traffic Measurement
- Doing the Math, Sizing • History of trunk requirements / Measure • 10 Trunks • G.711 Codec = 64Kbps ~ 90Kbps • 10 x 90Kbps = 900Kbps Sharing connection with Data? QoS to allow 70% of link for Voice = 1285Kbps • ITSP Supply that kind of DSL line? • Degradation further from Exchange • Multiple lines • Cable or Ethernet Services
- Quick SIP Trunks
- Internet ISP Data Asymmetric DSL TDM / PBX ITSP TDM to SIP/RTP Gateway SIP Trunks SDSL TDM PBX + SIP Trunks
- Data TDM / PBX TDM to SIP/RTP Gateway Voice Internet ISP ITSP Switch TDM PBX + Converged + SIP Trunks SIP / PBX
- It’s all about the ‘Edge’
- NAT SECURITY REMOTE USER QoS
- REAL-TIME LOW PRIORITYMED PRIORITY HIGH PRIORITY No ‘QoS’ on the Edge…! www @ $$$
- Packets Out SAP Email EF Default PolicePolice CBWFQCBWFQ Low Latency Queuing Packets In ‘Layer 3’ QoS Note: DiffServe Codepoint 101110 is recommended for the EF PHB. Hang on – 70% is your max! ClassifyClassify
- QoS implemented REAL-TIME LOW PRIORITYMED PRIORITY HIGH PRIORITY www @ $$$
- SESSION BORDER CONTROLLERS
- Which ITSP?
- PBX Router PSTN We need some more DDI’s Can I have a ‘London’ number? No problem Can we have our ‘old’ numbers please? Of course!That’s us PSTN ITSP Offerings
- PSTN ITSP Offerings PBX Router SHA / TLS / SRTPSHA / TLS / SRTP Thwarted We need more lines more our marketing push! I can do that via our Web Interface I’ll do that! PSTN
- Decide ITSP / Lists?
- Other Benefits
- 2211 PBX 2PBX 2 PBX 1PBX 1 Primary IP 55.43.123.11Primary IP 55.43.123.11 Alternate IP 55.43.123.12Alternate IP 55.43.123.12 CEO HQ PSTN DISASTER RECOVERY
- The All England Lawn Tennis and Croquet Club 300 – 400 Calls / Day Wimbledon 1000’s of people want tickets and information Increase Bandwidth and SIP Trunks Also add more SIP phones quickly Decrease after the tournament
- Stay On-Net!
- Multiprotocol Label Switching (MPLS) is a mechanism in high- performance telecommunications networks which directs and carries data from one network node to the next. MPLS makes it easy to create "virtual links" between distant nodes. It can encapsulate packets of various network protocols. MPLS
- MPLS Customer VPLS Site A Site C Site B PSTN
- ITSP Client PSTN You ITSP
- ITSP Client PSTN YouYou ITSP
- ENTERPISE TRUNKING
- SIP Connect
- PBX ITSP SIP Trunk Firewall/NAT
- That’ll be in 1.1
- Test Test test!!!!
- The road to compatibility Option Can May Should344344 475475 381381 144144 Source Alan Percy Audiocodes
- Oct 5th Oct 6th
- 16million
- RFC >150 Docs >1000 Pages
- NEW ONE PLEASE
- AT&T BBC Cable and Wireless Carousel Industries Cedarpoint CenturyLink Cisco Deloitte Ingate IPC Level3 Microsoft Mitel Mtsallstream Nacr Nec America Polycom Shared Technologies Telematrix Telquest Verizon Virtualhold West Windstream Xeta
- The Telecommunications Industry Association (TIA), the leader in advocacy, standards development, business development and intelligence for the information and communications technology industry, has officially endorsed the The SIP School as the provider of choice for training and certification for Session Initiation Protocol (SIP). 12 CECs Credits AT&T BBC Cable and Wireless Carousel Industries Cedarpoint CenturyLink Cisco Deloitte Ingate IPC Level3 Microsoft Mitel Mtsallstream Nacr Nec America Polycom Shared Technologies Telematrix Telquest Verizon Virtualhold West Windstream Xeta
- ITSP Checklist
- 1. Measure Calling patterns, in / out / busy times to measure requirements 2. Internal = G.729 – customer = G.711 3. Multiple sites and Redundancy 4. Select your SBC device or work with ITSP to decide 5. QoS on the Edge if voice / Data on same connection 6. Select ITSP and find out what is the interoperability of your equipment with them
- 7. Does ITSP have a lab to test re: PBX upgrades? 8. Do you have adequate security in place? 9. Take time to make change – it’s never smooth and business will get interrupted 10. Do/will ITSP offer advanced features such as Video, HD, Unified Comms etc. 11.Do you have an SLA or just a basic contract for service? 12. 90 day trial – test them out? Try all features – test their support team
- So I’m leaving am I? So I’m leaving am I?
- AT&T ask the Federal Communications Commission to create a timetable to shutdown the analog PSTN phone system in the United States. AT&T explains that “maintaining two networks - IP and PSTN is retarding the deployment of the newer broadband IP network”. [December 2009]
- Thank You www.thesipschool.com Code = ITEXPO2010 = 10%
Notas do Editor
- Welcome, my name is Graham Francis – I’m from The SIP School and thanks for coming along today “Understanding SIP because you’ll need to” is all about looking at what we’ve been used to such as analog / digital telephone systems and what we’re migrating to ~ VoIP and SIP networks … Why do you need to understand SIP? Well. It’s the foundation for all solutions but it may never be finished as a protocol so the ground is always moving under your feet so you’ve got to be ready But let’s start with the basics
- Now you are probably all aware of SIP and the fact that’s it’s pretty much everywhere now. it’s in the majority of VoIP products and services and it’s not going to go away, Game over – SIP has won!
- It’s in Phones Software Links UC Future – Smart Grids etc. etc. Future examples re: SIP forum new Smart Grid Surveillance cameras using SIP SIP on the Smart Grid
- But what about now, you want to implement SIP and see some benefits re: features and price
- Let’s focus on the most popular SIP service at the moment SIP Trunking What can you do to make sure it’s the best possible? What can you do in the situation of connecting your company to the rest of the world using SIP trunks? What questions do you need to ask of a Service provider?
- Now why the obsession with SIP?
- One of the most popular applications of SIP is SIP trunking where existing Basic rate and Primary lines can be replaced by SIP lines to bring about tremendous cost savings and you are going to see more and more of these services appear this year. And with people talking about 40 to 80% savings when using SIP trunks it’s easy to see the obsession.
- And here’s some more benefits of SIP Trunking, just to add to the obsession…. Buy SIP lines in any quantity you like. If you only need 10 lines that’s ok, you can always add more later If disaster strikes, your Main business numbers can be re-directed to an alternate location within minutes – maybe even sooner Direct Inward Dial DID numbers can be made available to all users in your company Call charges will be cheaper and maybe even no cost! SIP Trunks can backup existing E1/T1 lines until you are ready to switch permanently
- And just to highlight that point , have a look at this example… You can see clients using SIP to talk to their UC server for Presence status, this may send requests to the DNS to resolve SIP addresses to find SIP services. The UC server may then use SIP trunking to connect to some kind of gateway – maybe for Codec/Media translation to then allow trunking to a SIP based PBX from whoever. Don’t forget the SIP based IP phones and potential SIP trunking to messaging servers for Voicemail etc. And then the PBX uses SIP to connect to or traverse a SIP aware Firewall/NAT device that allows the SIP trunking to the ITSP Hey, I know there are other protocols in play on some networks but this is a SIP seminar so run with me here… You can see that SIP has become critical to a Unified Comms solution. But can you be sure it will all work?
- Regardless, you’ll need to look at the edge of your network
- Well, your selected ITSP should be able to provide you with Number Ranges, Move your existing numbers to their network, provide Emergency services support and Also have a good technical support team just in case you need help Features all there?
- Also, they need a good network infrastructure to ensure continuity of service… Give you a choice of Codecs to use and be able to provide a Secure connection to their network. And when you need more Trunks, can you manage your settings or do they do it for you? Security ITSP implement SIP security? RTP Security? SBC Decrypt after SBC? New Stack on PBX / IP Phones Firewalls etc.
- Flexibility in provisioning SIP trunks when demand grows is also a major advantage over traditional technologies. Take the example of the AELTAC club where they can manage 300 to 400 calls a day most of the year but during their Wimbledon tournament these rise to 1000's every day. By increasing Bandwidth and adding more SIP trunks during this period they can easily manage to provide service to their customers and once the tournament is over, they can reduce their Bandwidth, SIP trunk count AND their bill. How easy is it to do that with Analog or Digital lines? To read the case study kindly provided by Gamma Telecom in the UK in detail, please click on the logo
- Example – Level3’s network? For multiple offices and international presence Or small ITSP as one branch and local presence? Examples of what’s
- Multiprotocol Label Switching (MPLS) is a mechanism in high-performance telecommunications networks which directs and carries data from one network node to the next. MPLS makes it easy to create "virtual links" between distant nodes. It can encapsulate packets of various network protocols. Allows for full port speed No Committed Information Rate (CIR) = No discard-eligible traffic No Committed Access Rate (CAR) (i.e. MPLS over ATM/Frame) No need to manage PVCs & CIRs Automatically meshes the network Automatic re-routing via IP Ability to prioritize data applications Traverses over a Private-IP, not a cell-based network Access the corporate network remotely Reduces monthly network costs
- To build on the promise of cheap or free site to site calling it’s possible that you’d use an ITSP that has an MPLS network allowing you to have your own Private, Virtual Layer 2 network that gives you great connectivity and the opportunity to apply QoS rules to your Voice traffic. This is great but what you’ll find is that if you make calls to companies that are not part of this ITSPs MPLS network sooner or later you’ll have to connect to the PSTN.
- To build on the promise of cheap or free site to site calling it’s possible that you’d use an ITSP that has an MPLS network allowing you to have your own Private, Virtual Layer 2 network that gives you great connectivity and the opportunity to apply QoS rules to your Voice traffic. This is great but what you’ll find is that if you make calls to companies that are not part of this ITSPs MPLS network sooner or later you’ll have to connect to the PSTN.
- Xconnect – Neustar (SIPIX) Telcodrdia - Verisign
- Another organisation called The SIP forum are working hard to ensure (amongst other initiatives) that widespread adoption of their SIPconnect recommendation means that SIP Compliant products and services work when connected as most of the hard work has been done during the ratification process by the manufacturer and provider. So why not ask the people you’re working with if they are SIPconnect compliant or working towards it?
- Oh, but there’s a problem, that’s just version 1.0 Features like security, network call transfers, voice messaging, fax, hosted PBX functions, dealing with NAT and other areas were in many cases outside the 1.0 specification and thus not supported … Version 1.1 is in development so maybe the elusive ‘plug and play’ trunk is still some time away
- And as great as 1.1 sounds we really want 1.2 – now today…! The overriding goal of SIPconnect 1.1 is to correct errors and update protocol references in the ratified 1.0 specification. The principal focus of SIPconnect 1.1 continues to be the deployment of voice applications and the elimination of PRI interfaces between PBX systems and service provider networks. Though it is understood that Unified Communications will clearly be important in the future of SIP and enterprise communications, issues surrounding the introduction of presence, video, IM et al. into SIPconnect 1.1 are out of scope with the exception that service providers not interfere with end-to-end deployments of those applications.
- Now in order to do all of this testing you need to have an understanding of SIP and this is where education comes in……..
- But it’s still going to take some hard work – it’s still ‘early’ days for SIP! If you are a Manufacturer of SIP products, you’ve got to make SIP work as all your competitors are and if you leave it out you’ll be left behind. If you are a Reseller, you are the ones that clients will look to for intelligent solutions to meet their needs and run on their networks today AND tomorrow. Maybe you should look at your portfolio and see if the SIP products you sell will actually work with each other. Even put together bundles of tested solutions for clients to review… A PBX, A Firewall, A SIP Trunking service… etc. If you are a Client you have to balance the promise of all that SIP brings to the realities of making a solution work on your existing network…It should go without saying but, do not rush and please do your homework.
- Now we all hear that SIP has it’s problems and implementations may not meet the standard. The basic standard RFC 3261 is just to big and feature rich. I doubt that there is a single implementation out there which is 100% compliant to everything in 3261 The size of 3261 leads to the fact that every vendor implements only parts of the basic standard Thus the common feature set of all SIP implementations is very small, compared to the feature set of 3261. But on the other hand every customer obviously expects that the SIP implementation which he bought is 100% compliant to the standard … But whilst 100% Compliant to the standard may not be 100% compatible with other systems.
- SIP interoperability is VERY tricky because of the way that IETF Requests For Comments (RFCs) are developed. IETF RFCs and Drafts are developed in an open and communal environment, using committees and consensus to craft the specification. This has very many positive benefits, but also a few predictable negative side effects. The problem is that RFC 3261 that defines SIP has become "everything to everyone" and bloated in both size and in flexibility. Performing a simple word count on RFC 3261 yields some interesting insight into the problem:Weak Terms May = 381 Should = 344 Option = 144Can = 475 Strong TermsShall = 4Must = 631 As you can see, the number of weak terms "May," "Should," "Option" and "Can" outnumber the stronger "Shall" and "Must," which results in a very loose specification that allows the developers of SIP-based systems to make plenty of decisions on features of functions. The byproduct of this is that two systems can be completely RFC 3261 compliant and completely incompatible.
- So, how are you going to cope with the way SIP is and the problems and challenges that you may face? We believe that Education and understanding is the key for all…. Everyone in the industry needs to understand that true SIP interoperability is so far away that it may never be reached, so how do we all cope 1: Realise the facts in front of you 2: Education 2: Tell the Truth 3: Test until it all works and Client happy with all features. This sometimes means development along the way i.e. no Message waiting light in phones, No support for SIP Refer.
- Of course you could go to Google and type in “what is sip?” You’ll get around 15 million hits! So Happy learning there.
- You could check out the Request for Comment documents that define the protocol – over 150 documents with 1000s of pages – More fun!
- Or you could come and visit us at The SIP School The SIP School is the one place on the web to learn everything you need to get confident working with SIP What’s great about The SIP School is that it covers all the good stuff like the Basics SIP messaging The Servers Security – Firewalls – Nat – SIP Trunking Connecting to the PSTN Troubleshooting Enum services SIP and Unified Comms and so on. What’s really great about The SIP School is that it is updated as SIP evolves so people can keep their skills up to date – that’s a great benefit!
- And it doesn’t stop here – there’s more to come in the future… Ok, we’ve just added the module on SIP and Unified Communications …. Next up SIP and Mobile Technologies, SIP and the IMS, SIP and Cable, SIP and IPv6 etc. etc. etc. So one place is all you need and that’s the good news for anyone who needs to work with SIP.
- Now we have a lot of friends using our system already, including some pretty good endorsements of our training and SSCA Certification. And today I can tell you about our new endorsements which in effect make us the leading SIP training and certification company – worldwide.
- Now we have a lot of friends using our system already, including some pretty good endorsements of our training and SSCA Certification. And today I can tell you about our new endorsements which in effect make us the leading SIP training and certification company – worldwide.
- (Call Centre retaining user info?) Internal transfer doesn’t go back to ITSP and use another trunk + lose caller information centrex line is feature rich POTS line. Centrex lines offer customers numerous services including caller id, call transfer, three party conferencing, ring back, call group hunting, call pickup, Costs: Carriers appear to be charging the same per minute for both solutions. The potential cost savings are in the access, lower power consumption, and equipment. The business case for SIP Toll Free Trunking will most likely need to be business value driven vs. cost savings driven.
- Now SIP trunks are aiming to replace the PSTN, but the thing is, it’s going to take a long time and it’s usually the PSTN that bridges all of these Voice over IP based islands that are popping up now. I think it will take a very long time for the PSTN to become useless and in fact it’s going to stick around as a pretty good backup if the SIP Trunks go down.
- The road to successful deployment of SIP trunking across a company can be long hard and fraught with difficulties. But the more you know about SIP and SIP trunking and the more direct you are with ITSPs with your requirements, you can take some of the bumps out of the ride.
- Thanks for your time Now if you have any questions let’s see if I can help out…