Andrew Prokop has been heavily involved with SIP and VoIP since the late 1990’s. He holds six United States patents in SIP technologies and was on the teams that developed Nortel’s carrier-grade SIP soft switch and SIP-based contact center. Through customer engagements, users groups, tradeshows, and webinars Andrew has been an evangelist for digital transformation for enterprises and their customers. Andrew understands the needs of the enterprise and has the background and skills necessary to assist companies as they drive towards a world of dynamic and immersive communications.
Hi Andrew, I’m following blog daily. if your blog have one Index page it will be very easy to Navigate to all your posts, Because your each every post is Very Useful. so for Beginners like me can Navigate easily from Basic to Advanced level Sip Concepts. one again Thank you for posting such useful articles.
Thank you. That’s a worthwhile idea. I just need to find the time! My blog is just one of the many things I do during that day. It’s hard enough to write the articles. 🙂
I will think of what I might be able to do, though.
HI Andrew ,
While doing a google search for “traceSM” , I reached in your site. It is a simple but quite interesting blog.
My work is primarily focused on the product in which you designed , Avaya Aura Contact Center.
I enjoyed reading and looking forward for more posts ..
Just a quick thank you for your excellent blog! I work for an Avaya BP in the Netherlands and often refer to your blog when training junior engineers. Please keep it up as understandable SIP information is often hard to find!
Thank you! I really appreciate this.
Andrew, it would be interesting to learn from you about Fax over SIP
How about this: https://andrewjprokop.wordpress.com/2013/06/23/fax-over-ip/
Can you please clarify SIP OPTION timeout handling on IMS core?
If UA-B is unreachable, should S-CSCF send 408 timeout response to originator UA-A based on T2 timer?
RFC 4321 states not to use 408 Timeout.
I am not sure what would happen. Perhaps a 500 error. The sending UA’s SIP stack could also simply timeout and kill the transaction.
Hey Andrew, as a fellow ex Nortel employee, it’s good to see that life does not end after it’s demise, though it seemed like it at the time. I worked as a Post-Sales Customer Engineer for DMS, then right through its Succession journey to CS2K and ended up latterly in CS1K.
While now working for a large Insurance company (Avaya Aura shop) as a Voice Architect, I was digging into some SIP call flows in traceSM the other day, i was surprised to notice that the SIP User agent used in SM is the old NGSS code that I remember from CS2K.
It’s a small world.
Love your blog, keep it up.
Hi Andrew, Thank you for all the information you put for us on your site. I am just a simple SIP user and far from being an expert. I hope this is the right place for asking my question about re-invites in SIP. If not, I apologize in advance and maybe you can redirect me to the proper channel.
My question is simple: I understand that a re-invite get initiated after the ACK, but do you think is there a way of having a re-invite before answering the call (during call progress for example)?
You can use a Re-INVITE before the ACK and I have seen that happen, but mostly you would use an UPDATE.
I had been following your blog on many technological domains, while I was going thru some of the Session manager, SBC and SIP Articles of yours, which were well explained in short paragraphs. I want say Thank you for your good articles.
Just want to let you know, you have an awesome style for explaining complex things in simple manner. Your blog is an inspiration for a passionate technical writer like me. I have been working in VoIP and Telecom industry for last 6 + years, and cant tell you how enjoyable your blog posts are!
Stay blessed and keep writing the great stuff,
Entered to VOLTE/IMS world 2 years back, but many fundamental SIP headers like AOR, CONTACT header , PATH header, TimerB vs T1 etc. etc. were not 100% clear to me. After I found your blogs on these( still going through rest in my own pace) I am now 100%clear with the usage of each SIP headers.
Thank you for sharing…., much appreciated.
Hi Andrew. I have a question about SIP. Is it possível to send a INFO message at any moment of the call or it’s necessary to wait for the ACK? For exampe. INVITE, 200ok, ACK and INFO or INVITE, 200ok, INFO, ACK. I am asking you because one vendor says the first one is correct and the other the second one option. By the way the softswitches are using SIP-I. Thank you.
I believe that as long as the To and From tags have been established, INFO can be sent before ACK.