Dogtel IT » VOIP

SIP three way handshakes

McCool — Tue, 31 May 2016 05:08:55 +0000

Time for another post on SIP, why SIP? Because SIP is good, SIP is like a nice dessert, it is something that just works. This time I want to touch on some of its basic fundamentals; the three way handshake. I decided on this topic because recently I have been working on a number of issues that had to do with incomplete SIP messaging between endpoints and calls that were being cut off after about 30 to 40 seconds. Now, when you come across calls that disconnect after a certain and set period time and time again, you can bet money on it, that some timer has expired forcing the call to drop.

So let me brush up some basics regarding SIP call establishment. SIP Sessions are used with Voice and Video over IP, to establish a call session between users. The core SIP specification provides a way to set up and manage sessions between two user agents (i.e. endpoints, such as desk phones, or software SIP clients).

SIP sessions, sometimes referred to informally as “calls” and more formally referred to as dialogs, are created via invitations from one User Agent (User Agent Client or UAC) to another (User Agent Server or UAS). This invitation transaction is basically a three-way handshake between the UAC and UAS that consists of the initial INVITE message sent by the UAC to the UAS, one or more provisional responses sent by the UAS to the UAC, a final response sent by the UAS to the UAC, and an acknowledgment sent by the UAC to the UAS.

The three way handshake (similar to a TCP three way handshake) in SIP works as follows INVITE –response (1xx) – ACK. So as soon as the INVITE is sent to request a dialog (Alice in the example picture above), Bob sends a 100 TRYING status update back, so UA1 does not re-transmit its INVITE. The 180 RINGING is sent back to Alice to signify that the phone is ringing. The 200 OK is sent from the called party as soon as Bob picks up the phone (in this cased the OK also contains the SDP). After this the ACK is send by the caller. Remember the ACK is always sent by the party that sent the initial INVITE. This ACK as part of the 3 way handshake is important because with this ACK the caller confirms back to the called party that it has received a final response to an INVITE Request. All these messages within the three way handshake will have the same Cseq # in their message header.

So, in a nutshell, the 3 way SIP handshake is:

The caller sends an INVITE
The callee sends an 200 OK to accept the call
The caller sends an ACK to indicate that the handshake is done and a call is going to be setup

I you want more in depth documentation on SIP, i suggest you read RFC 3261

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SIP trunking; where are we at now?

McCool — Wed, 17 Jun 2015 06:10:10 +0000

SIP Trunking; where are we at now? Chasing a myth?

SIP is rapidly becoming the de facto standard protocol for IP telephony. Hang on, did I just answer my opening question? Well not quite, so let me re-phrase that question and see if I can find some insight into the fact that, although SIP is increasingly pervasive, then why is it that ISDN is still the preferred way of connecting our PBX to a carrier?

SIP is a relatively long protocol. Its initial standardization took place in 1999 and SIP 2.0 was only published in 2.0. SIP was originally developed based on voice applications, but has always been envisioned to be extended beyond that, and indeed, apart from telephony (SIP dialing) it currently supports, video conferencing, streaming media distribution, instant messaging, presence information, file transfers and much much more.

Straight away you can see how versatile SIP is, compared to its alternatives such as ISDN, MGCP, H323, QSIG and Skinny. All these protocols, apart from H323, are traditionally used to exchange telephony capabilities and set up telephony calls. ISDN is fastly becoming old school, yeah sure it is a stable and proven technology, that offers circuit switched connections, using copper infrastructure, but it has some serious limitations. Video and ISDN? Not a hope in hell, unless channel bonding is your thing. Addressing and ISDN? Same as in the days of yester year; phone numbers. There is no IP in ISDN. ISDN has been around since 1988 and will not get much older. Increased fibre roll out in many countries, will only speed up this process. In Australia, the catalyst for killing ISDN dead, is the NBN roll out. So this pretty much rules out the future of MGCP as well (not that it ever was a real Telco grade protocol anyway).

So you would think that SIP and H323 would go head to head to take over from ISDN. Well not exactly. H323 is on its way out as well. The question is; ‘why?’. H323 has its foundation as a multimedia protocol and SIP has its origin as a VOIP communications protocol. So these protocols are to a large extent similar in their capabilities. I feel the biggest difference and benefit that SIP has over H323 is its extensibility. Sure H323 is somewhat extensible, either in standardized form or vendor specifically, but no where near as substantial as in SIP.

You could argue that there is nothing that SIP already does, H323 cannot do. Then why doesn’t H323 do so?

Globally Telephony providers are obviously slowly starting to market SIP trunking as the alternative over ISDN. A lot of critisism regarding SIP is aimed at its inter vendor compatibility, which people claim is troublesome and complex. But really, compatibility problems between vendors are inherent to every protocol and SIP is no exception.

You could say that the protocol battle for dominance in the IP Telephony space has been won in favor of SIP, irrespective of one’s belief that it is the best protocol for the job.

Further migration of ISDN to SIP is highly dependable of further fibre network expansion

So, why, if SIP has been around about 16 years, is it that it is still not fully pervasive? Well, in the PBX space it already is, Cisco use it, Avaya use it and so does Nortel just to name a few. It the PBX connection to the service provider (read PSTN) that is only slowly becoming true IP. The reason why the adaptation to SIP is so slow on this front, is not because SIP is not a good solution, it is because there was no real drive from the providers to make the transition to SIP.

Telephony providers have, in the past, been reluctant to really prominently put SIP forward as a service. Mainly because of fear of cannibalization of their highly profitable ISDN services. And in addition to this, businesses were lacking the technical understanding to consider SIP trunking as a viable option for ISDN replacements, so logically there was very little commercial drive for the providers to expand and invest in SIP Trunking.

Because ISDN usage penetration is so high, it would mean that replacing it with SIP and essentially converging voice and data would mean major investments by these same providers. Of which the biggest investment would be to put glass fiber infrastructure into the ground, as copper is simply not future proof, and I am not even going to argue why that is. This change and investment in underlying infrastructure will truly converge voice and video, and will thus make SIP more prominent. In Australia the NBN will do just that. Now I am not arguing that the NBN roll out is a panacae for all that is wrong with our current copper infrastructure. What I will argue is that it will have a drastic impact on ISDN, in the way that it will end up as a wayside technology. In the same way as coaxial Ethernet and dialup modems have found their grave; because they were simply superseded. You might still not like NBN, well get used to it. New residential areas and industrial estates around Australia are fitted out with fiber not copper. This means that even if you order ISDN in these areas, what you are getting is emulated ISDN being presented to you by some provider’s IAD device and you plug it into your ISDN capable PBX. So emulated ISDN over the same fiber strand that goes into your premise and that also carries your data. Bye Bye separate circuit, bye bye redundancy.

So where are we at now? Well SIP communications will continue to grow. How fast this growth will be, depends on the cost model that the providers will use. In other words, until it becomes appealing for organisations with existing ISDN lines to migrate to SIP trunking. Combined with the further roll out of fiber infrastructure.

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VOIP, what’s the fuzz all about?

McCool — Tue, 01 Jul 2014 08:22:46 +0000

Dogtel can implement VOIP in the Melbourne area and we are very passionate about it, so let us explain about its advantages.

VOIP has been around for a while. People also use terms like IP Telephony and Internet telephony (made famous by companies like Skype). But let’s stick with the term “VOIP”.

The Telephony landscape is split up in traditional telephony on one side and VOIP on the other. Traditional (PSTN) telephony has been around since Alexander Graham Bell took credit for its invention in 1876. It is very robust and obviously has a long proven track record.

VOIP, on the other hand, has “only” been around for over a decade, but has increased in popularity in the last decade. In fact, in 2008, 80% of all newly implemented phone systems (PBX’s) were VOIP. The main decider for larger companies to go VOIP, is the fact that VOIP can be run over the same, existing, network as Data traffic, significantly cutting costs. This compared to traditional PBX’s, that physically run completely separate from all the other data traffic. Consider an standard office with desks that have at least two outlets available; one for a telephone and one for a telephony. It is not hard to understand that both networks exist side by side substantially increasing the cost to upkeep them.

Consumer market voip melbourne

The growing popularity of VOIP in the consumer market is mainly contributed to, by the increased availability of broadband internet. Service offerings like “Naked DSL” and unlimited domestic calling for a flat monthly subscription fee are examples of this.

A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways:

Dedicated VoIP phones connect directly to the IP network using technologies such as wired Ethernet or Wi-Fi. They are typically designed in the style of traditional digital business telephones.
An analog telephone adapter is a device that connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. Some residential Internet gateways and cablemodems have this function built in.
A softphone is application software installed on a networked computer that is equipped with a microphone and speaker, or headset. The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input.

With the options above a VOIP phone user can have the same user experience with a VOIP phone, as with a traditional phone. It also means that with a VOIP service provider, faxing is easily integrated into it.

Corporate market

As stated earlier, medium and large businesses have been moving away from traditional copper wire telephony to VOIP, not only in the Melbourne area, but around the world. And for good reason.

Because VoIP allows both voice and data communications to be run over a single network, it can significantly reduce infrastructure costs.

In addition to this, companies that are spread across multiple locations, can make huge cost savings when directing their calls across their existing inter office Data connections.

In the corporate market, the term “VOIP” has evolved into unified communications services that converge all communications—phone calls, faxes, voice mail, e-mail, Web conferences, and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Many major software and hardware vendors such as Cisco and Microsoft have embraced Unified Communications and can offer its integration with their existing products such as email, Active Directory and network components and protocols. The convergence of voice, video, and data communications around a shared IP-based infrastructure allows users to easily make a call, send a message, or join an audio or video conference.

To sum up some of the advantages that VOIP has over traditional Telephony:

VOIP easily supports Tele-workers (home office becomes extension of regular office)
Cost savings by converging the Telephony and Data Communications networks
Wide range of choice of VOIP providers
Offers call cost savings for organizations with multiple locations
Supported by well know vendors.
Ease of number or number range portability and office move
Unified communications allows rich content sharing, combining it with video and audio.

We at Dogtel are completely sold on VOIP and have extensive experience in the field. VOIP Melbourne is a competitive market, but we know it well.

Talk to one of our consultants and Let us help you improve your IT.

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