Rummaging Through the SIP Closet

I have a difficult time throwing things out.  

Perhaps it’s due to my optimistic nature, but I always feel that no matter how long something has sat around unused, there is a good chance that it will prove useful sometime in the near future.

Just take a look at my dresser drawers and closets. They’re filled with shirts, pants, coats, suits, and ties that I haven’t worn in years. Some are practically threadbare and some show little signs of wear. The condition isn’t even a deciding factor, though.

I keep things because, well, just because I keep things.

SIP has a little of that, too. While it hasn’t been around as long as some of the t-shirts I own (I am not kidding), there are a few relics that deserve a second look.

This article originally appeared on SIP Adventures and is reprinted with permission.

For the next few minutes let’s take a journey through the closet of SIP and see what, if anything, can be tossed or at least donated to the Goodwill.

As you know, headers are used to identify the characteristics of a session.

For instance, To indicates the destination of a session and From identifies the creator.

There are many more headers that you see on a regular basis in both requests and responses. However, there are several headers that in all my years of SIP programming and Wireshark traces, I’ve never come across.

So, in no particular order, let’s take a look at a few of them.

Related article: Wow, I Can Do That with Unified Communications?

The first on my list is Subject. Like the subject of an email, this header can be used to provide a brief reason as to why the session exists.  For example, I might tag a phone call as “lunch” if I were calling to invite a coworker out for a quick bite.

A few weeks ago, I wrote about providing the context of a call in my article Contextual Communications. Subject would be perfect for that.

You might want to answer the call tagged with the subject “Quarterly sales numbers” over the one labeled “My vacation to Akron” — or vice versa. A little extra context to phone calls would go a long way in terms of prioritizing your day.

Subject: I am calling about today’s meeting

However, as cool as having a subject associated with a phone call is, I have never actually seen it implemented. There is nothing on any of my hard, soft, or mobile SIP phones that allows me to enter a subject. Even if such a parameter existed, none of my devices have a designated spot to display it.

A perfectly good header that provides a very cool feature sits unused.

I can say the same thing about the Priority header. Similar to an email priority flag, you can set different levels of session priority. So, instead of just a subject that says “Quarterly sales number,” you could also tag it as “High Priority.”  This adds even more context to another otherwise uncategorized call.

Sadly, I have never seen the Priority header used, either. Would I love to click a priority flag on an outgoing call? You bet I would. I would love the recipient to know when he or she can safely ignore my call or when it absolutely needs to be answered.

There are a few more headers that fall into the “defined, but I have never seen” camp. I can find uses for the Alert-Info and Organization headers, but sadly, I’ve never seen a SIP device that supports either one.

Next are the Response codes.

As with a few SIP headers, there are response codes that I’ve never come across in all my dealings with SIP. It’s not that I think they shouldn’t exist. I just can’t seem to find anyone who has actually implemented them.

We should all be familiar with a 200 Ok response.  You send or receive one of those every time a call is answered. (If you read my article, REFER Revisited, you should also understand 202 Accepted).

Who out there is familiar with the 204 No Notification response?

The SIP specification says that this response indicates that “the request was successful, but the corresponding response will not be received.” Huh? I’ve poked around the Internet and found a document that says it will only be sent after a SUBSCRIBE, but it was completely vague as to when and why. It might have a wonderful use, but until I find a better explanation, it goes into my “Why is this here?” pile.

I am sure that some of my readers have their own list of unused, little used, or “huh” aspects of SIP.

However, like my drawers filled with old race t-shirts, I expect that that they will be around for quite some time. After all, you never know when you just might need something.

If you’re like me, it’s probably a couple days after it has been tossed out.

Related Articles:

How to Realize Huge Cost Savings for Enterprise Telecom

This case study shares the steps taken by Avaya to reduce a sizeable telecom bill by 50%. The methods are trending tactics used by a wide range of organizations, and in combination have proven to produce especially favorable results. Keeping in mind that most IT operation budgets are flat, cost savings of this size can give your organization a greater opportunity to invest in innovation.

When every penny counts, internal telecommunication expenses must be closely accounted for. It’s a large expense and pain point for many organizations, and adopting change in this area is often associated with fear and uncertainty.

In reality, these fears disappear with a pinch of planning and excellent execution. Prioritizing telecom costs as a business initiative, establishing expectations, and determining a reasonable timeframe are some of the leading difficulties in taking on a project of this magnitude. These steps can have a great impact on the overall success of the project.

  1. Hire an expert to properly analyze your costs.

    There are many nuances in telecom, so if your organization does not have an expert internally, then it would be worthwhile to hire a third-party consultant to analyze the cost. They can provide better financial estimates of the cost and savings to ensure adequate ROI.

  2. Centralize trunking.

    To make it simple, paying for fewer circuits reduces cost. At Avaya, we reduced our total internal global telecom costs by 50% and got a 40% overall reduction of IT operational spending by doing so. Pay close attention to the amount of money paid in relation to the locations and compare centralized trunking cost to your current expenses. There are also considerations based on regional laws—for example, telecom laws in India may prevent this practice altogether.

  3. Reduce local trunking.

    Gartner estimates that “network architects and procurement managers can leverage SIP trunking services to slash enterprise telecom expenses by up to 50%.” As we all know, telecom does not come cheap, so these are results that cannot be ignored. Since IT operating budgets are often flat, this method can be a valuable way to extract savings from existing expense.

Gartner further suggests that, “enterprises should leverage the competitive SIP trunking market as U.S. service providers are reducing rates to win new business and retire their legacy TDM networks.” Needless to say, if you have yet to board the SIP train, you should strongly consider doing so now.

Previously at Avaya, all locations had their own local trunking and now we are saving cost by routing calls through a central location. In fact, SIP trunk consolidation provided an average savings of ~40% per month over PSTN trunks. We did it by implementing Avaya Aura to connect sites between a WAN used for both voice and data.

The Benefits Speak for Themselves

Consider what your company could do with these savings. We chose to reinvest 20% of our savings towards innovation and new capabilities. Gaining support from your colleagues should be a bit easier with these proof points in your back pocket and a strategic plan of action.

Understanding SIP PRACK for Avaya Aura

As many of my readers know, every few months I teach a two and a half day class on “all things SIP.” My students are exposed to everything from “why SIP” to the nitty-gritty of SIP requests, responses and call flows. I even speak about some of the more esoteric topics such as To and From tags, the Replaces header, nonce values and TR-87.

Included in the esoteric list is the PRACK (Provisional Response Acknowledgement) method. PRACK wasn’t in the original SIP specification and was introduced later in RFC 3262. It came about after it was realized that some user agent servers need to know that a provisional response was received by a user agent client. Before PRACK, 1xx responses sent using UDP might get lost, and the sender would never know. PRACK adds a layer of reliability to an otherwise unreliable call flow.

I previously addressed PRACK in my article “Ducks Go Quack. SIP Goes PRACK.” Although I addressed most of the pertinent material, I was short on examples and real-life call flows. As I walked my most recent students through live calls on my company’s Avaya system, I happened to notice a few PRACKs and decided it was time to update my old article.

The following screenshots were gathered using the Avaya traceSM utility. I simply started traceSM on a live Aura system, let it run for a few minutes, and then stopped it after I noticed a few PRACK messages fly by. This was simply because I was unsure as to when Avaya uses PRACKs and when it does not.  In other words, “When in doubt, trace it out.”


Let’s start at the beginning. PRACK messages aren’t just sent out-of-the blue. The sender of an INVITE message must indicate that it is capable of sending PRACKS. It does that by including the header in the INVITE message:

Supported: 100Rel

This tells the recipient that, if requested, it will send PRACK messages for 1xx Responses.

The following shows an INVITE with such a header.


Now that the user agent server knows that PRACK messages are possible, it will include headers similar to the following in all 1xx Responses it wants to be PRACKed:

Require: 100Rel

Rseq: 1

The Requires header with a value of 100Rel tells the user agent client (the sender of the INVITE) that a PRACK is expected for this response. It’s important to know that the user agent server (the sender of the Response messages) has to request the PRACK. It’s not an automatic process and must be initiated with an Rseq header.

The value in Rseq is used by the user agent client when it creates a PRACK message. The user agent server is responsible for setting and incrementing this number.

The following 180 Ringing indicates that it expects a PRACK.


Upon receipt of this 180 Ringing, the user agent client must respond with a PRACK message. Of interest to this article is the Rack header. This header must contain the Rseq value sent in the previous 180 Ringing. Additionally, it will indicate the original INVITE session’s CSeq number. Look back at the INVITE in this call flow, and you will see a CSeq value of 1 (one). Therefore, the Rack will look as follows:

Rack: 1 1 INVITE


Next, the user agent server will send a 200 Ok for the PRACK. This tells the user agent client that the PRACK was received and processed.


For grins, I will now show you the 200 Ok for the original INVITE. Note that it does not have a Rseq header and 100Rel is not in the Requires header. Why not? That’s because this is not a provisional response. PRACKs are only sent for 1xx responses.


Mischief Managed

Before I close things out, I want to address the question I hinted at near the top of this article.  When does Avaya use PRACK?

While I honestly don’t know all the permutations, it appears that an INVITE from an Avaya endpoint will always indicate that it supports PRACK (Supported: 100Rel).  However, as you just learned, it’s the recipient of the INVITE that indicates if PRACK messages are required.

In the example above, the Avaya Modular Message voice mail server requests PRACK messages.  Additionally, PRACK is used when direct media is enabled.

There is a good chance that PRACK is used in other situations, but I am going to have to start up a few more traceSM sessions to learn where they show up.

That’s about all I really need to say about PRACK. I invite you to take a look at the RFC if you want to learn about any PRACK subtleties I might have missed, but for all practical purposes, I’ve said all that needs to be said. I hope you had as much fun today as I did. As is often the case, I learned something in the process of writing this article, and that’s always a good thing.

Understanding Avaya Aura SIP Registration

“Let’s start at the very beginning/a very good place to start/when you read you begin with A B C/when you sing you begin with Do Re Mi.”

I have always loved musicals, and Rogers and Hammerstein’s “The Sound of Music” is high on my list of favorites. Sure, it’s corny and far from historically accurate, but that doesn’t bother me in the least. I’m always willing to set aside any sense of reality for good singing, romance and adventure, and “The Sound of Music” has them all.

So … what does this have to do with unified communications? REGISTER, of course. Like Do Re Mi, you begin SIP with REGISTER.

This article is a continuation of the concepts I presented in A Close Look at Avaya Aura IMS Call Processing and An Even Closer Look at Avaya Aura IMS Call Processing, and I’d suggest you take a look at those before tackling this one.

Can you get SIP devices to communicate without REGISTER? Absolutely. In fact, when I teach my SIP class, the students put their SIP clients into point-to-point mode, which does not require REGISTER. This means that clients send SIP requests and responses directly to the other clients, not through a proxy. The clients can do everything all by themselves.

However, point-to-point without REGISTER has a serious downfall. The clients are required to know the IP addresses of all the other clients they wish to communicate with. While this is fine in a limited classroom environment, it becomes unwieldy after you grow beyond a handful of endpoints.

As an analogy, imagine having to know the IP address of everyone you wanted to send an email to. That’s the same problem you have if you don’t use REGISTER. It’s simply not practical.

The Tie that Binds

REGISTER associates a user’s identification, or Address of Record (AOR), with one or more locations. Note that I said locations. You are not limited to registering an AOR to a single device. Personally, I routinely register my AOR to a physical desk phone and multiple SIP soft-clients. Avaya Aura supports up to ten such registrations per user. That’s enough to make even the most device-crazy nerd happy.

You bind an AOR to an IP address with a Contact header.  For example, one of my soft clients might tell a SIP registrar that aprokop can be reached at with this Contact header.

Contact: Andrew Prokop <SIP:aprokop@>

Registrations are time-based and will eventually expire. This requires the client to periodically refresh a REGISTER with a new REGISTER. Actually, new isn’t the correct word to use for this. Subsequent REGISTER messages must contain the same Contact, To, From, call-ID and From Tag as the original registration. This allows the SIP registrar to know that it’s simply a refresh and not a new registration for the same AOR.

Note that CSeq will increment with each REGISTER sent.

Keeping Things Secure

I might tell my communications system that I am Andrew Prokop, but it would be foolish to trust me at face value. That’s why SIP allows a REGISTER to be challenged.

Before I go through a REGISTER challenge, allow me to define something known as a nonce.

Nonce stands for Number Once and is an arbitrary number used only once in a cryptographic communication. The recipient of a nonce will use it to encrypt his or her credentials. Number Once refers to the fact that encryption with this nonce can only be done one time. If someone were to sniff the LAN and obtain someone’s encrypted password, it won’t do them any good because it can only be used in a single transaction. It becomes stale and useless immediately after its first use.

A REGISTER flow is fairly simple and follows these steps:

  1. A user sends a REGISTER to the SIP registrar. For Avaya Aura, this is a Session Manager. The To and From headers contain the user’s AOR. The user specifies the number of seconds the registration should be valid in the Expires header. This value can be later raised or lowered by the registrar.
  2. The registrar returns a 401 Unauthorized response with a WWW-Authenticate header.  This header contains data that must be used to encrypt the user’s communications password. Specifically, it contains a nonce along with the name of the encryption algorithm that the client must use.
  3. The user sends a second REGISTER to the SIP registrar. This REGISTER contains an Authorization header. Within Authorization is the user’s encrypted password.
  4. If the correct password is received by the registrar, a 200 Ok response is sent to signify a successful registration. An Expires header may be present with a different value than what the user requested. This is the time the registration will be valid as determined by the registrar’s policies.

A registration is removed by sending a REGISTER with an Expires header value of 0 (zero).

In a picture, we have this.

Reg1Using the traceSM tool on an Avaya Aura Session Manager, I captured the following trace that shows a REGISTER, the challenge and a REGISTER with encrypted credentials.  Take a look at the headers, and you’ll see that they’re doing exactly what I said they would do.

Reg2 Reg3 Reg4


In the case of my daily work life, my various SIP devices will each send a REGISTER, be challenged and resend the REGISTER with the encrypted credentials. They periodically refresh their registrations to ensure that I am able to make and receive calls on all my devices until I am finished for the day.

Speaking of finished for the day, that’s about all I have to say about REGISTER. It’s not that complicated once you understand the basics. Just keep in mind that while registration isn’t absolutely mandatory, it enables a secure, scalable and easy to manage SIP solution.

… And these are a few of my favorite things!

Andrew Prokop is the Director of Vertical Industries at Arrow Systems Integration. Andrew is an active blogger and his widely-read blog, SIP Adventures, discusses every imaginable topic in the world of unified communications. Follow Andrew on Twitter at @ajprokop, and read his blog, SIP Adventures.