Some really good stuff in here and is especially helpful to start to understand P-Asserted and P-Preferred headers which is necessary in the UK as Ofcom is stamping down on number spoofing. The insertion of lines and highlighting of text (around 20:00) is an excellent technique but my goodness I was getting seasick trying to follow. Lots of pauses, rewinds and playing at .75 got me through!
That is a good question to know the answer to! You can think of a phone call as having two phases. The first phase is the signaling phase. This signaling phase is where the SIP protocol operates. In the signaling phase, the originating device sends the call setup request to a call agent like CUCM and the call agent determines what to do with the call. The call agent will then use a signaling protocol to reach out to the target. This second signaling leg can be the same signaling protocol (SIP) or another signaling protocol (SCCP, MGCP, H323, etc.). Once the two signaling legs are established it is time for phase two of the call, which is the payload exchange phase. In this phase, the call agent uses signaling to tell each of the two endpoints in the call about the IP, port, codec, DTMF, etc. that it should use to send RTP (which is the payload exchange protocol) to the other endpoint. At that point, the two endpoints exchange RTP with each other directly to carry the voice or video payload. Which, I suppose doesn't exactly answer your question. Here is another way to think about it. It is possible for two endpoints to signal directly to each other to set up a call and then exchange RTP directly as well. We add a centralized call agent into the mix for signaling so that we can have a centralized dial plan, enforce calling privilege policy, call admission control, and other centralized call control features. The RTP stream, on the other hand, does not need to flow through the call agent since the call agent's job is to only handle the signaling. I hope this answers your question. If not, please let me know. Maren Mahoney - Sunset Learning Institute
Hi Dixon, Make some SIP calls. Trace them. All SIP devices generate these messages. If you do not have access to SIP devices, there are many websites that break down SIP messages. Here is a great one: www.3cx.com/blog/voip-howto/sip-messages/ -John
Hello, we're glad you found this helpfull! We have several resources like this within our neXT Learning Community portal. It also includes a discussion board to chat with our instructors. If you are intersted, you can email neXTLive365@sunsetlearning.com and we will get you access into the community for free!
Hi Prashant, I am not exactly sure which product you’re asking about. There are two that come to mind: Cisco has a vCube - a virtual CUBE that runs inside of their UCS/VMWare servers. This is a something you would use in a production environment, instead of using an entire physical router. The other product you might mean is GNS3. This is used for self-study and practicing learning the commands. - someone who doesn’t not need a real production CUBE. GNS3 is software that you can run on our laptop, it allows you to install and work with a real router operating system (IOS) assuming you have one available to use. In my video, I was using a physical 2911 router/CUBE via telnet. JOHN MEERSMA
This is one of the amazing and most thorough video I have seen regarding SIP. Thanks for the tutorial.
Thanks so much! We're glad you enjoyed it!
this is a fantastic learning tool. You have made this so easy to comprehend! I am only 8 minutes in and have learned 100 things!
Wow, thanks so much for the feedback! So glad you found it helpful!
Exccellent. You have done a beautiful work explaining the sip call stack. Thank you
Glad you enjoyed it!
This is gold.
Awesome video, best i've seen on understanding SIP messages thanks john.
Thanks for the feedback, we are glad you found it helpful! You can find more videos like this here: www.sunsetlearning.com/tag/cisco-collaboration/.
Its an excellent tutorial. The way of explanation is amazing. Appreciate if you provide videos for rest of the topics as well.
Thanks for the feedback! We're working on some exciting new content now!
Some really good stuff in here and is especially helpful to start to understand P-Asserted and P-Preferred headers which is necessary in the UK as Ofcom is stamping down on number spoofing. The insertion of lines and highlighting of text (around 20:00) is an excellent technique but my goodness I was getting seasick trying to follow. Lots of pauses, rewinds and playing at .75 got me through!
yes, it's definitely a lot! Thanks for watching!
Excellent tutorial. This is what I have been looking for to get minute details in SIP traces. Though SIP is an ocean subject, I got some idea.
A very informational and well explained video !!
Excellent video!!!
Excellent ! I am new to SIP, I found the tutorial very informative.
have u joined in sip course?
you are the only one make it as simple class
Thanks for share. its pretty informative for a beginner. I kinda developed interest to learn SIP in more detail.
Great! Thanks a lot for sharing.
Glad you enjoyed it!
The explanation is too good 👌👌👌👌👌👌
We're glad you found it helpful!
Excellent tutorial.. Very useful
Thanks for watching!
Awesome Video....extremely helpful.
Very informative.Thanks a lot.
Thanks for the feedback!
Gr8 .. very Useful. Thank you so much!
Noob question. Why does SIP has to pass through a server/CM but RTP can go straight to its destination?
That is a good question to know the answer to! You can think of a phone call as having two phases. The first phase is the signaling phase. This signaling phase is where the SIP protocol operates. In the signaling phase, the originating device sends the call setup request to a call agent like CUCM and the call agent determines what to do with the call. The call agent will then use a signaling protocol to reach out to the target. This second signaling leg can be the same signaling protocol (SIP) or another signaling protocol (SCCP, MGCP, H323, etc.).
Once the two signaling legs are established it is time for phase two of the call, which is the payload exchange phase. In this phase, the call agent uses signaling to tell each of the two endpoints in the call about the IP, port, codec, DTMF, etc. that it should use to send RTP (which is the payload exchange protocol) to the other endpoint. At that point, the two endpoints exchange RTP with each other directly to carry the voice or video payload.
Which, I suppose doesn't exactly answer your question. Here is another way to think about it. It is possible for two endpoints to signal directly to each other to set up a call and then exchange RTP directly as well. We add a centralized call agent into the mix for signaling so that we can have a centralized dial plan, enforce calling privilege policy, call admission control, and other centralized call control features. The RTP stream, on the other hand, does not need to flow through the call agent since the call agent's job is to only handle the signaling.
I hope this answers your question. If not, please let me know.
Maren Mahoney - Sunset Learning Institute
Thanks a lots for clearing my doubts
We're glad you found it helpful! Thanks for watching!
Can you please share the Slide Deck shown in the presentation
Unfortunately, we don't have the slide deck available. Sorry!
amazing tutorial, best ever. thanks a lot
Great Great video! You wouldn't happen to have your slides and output saved somewhere, to download would you?
Very well explain
Hi sir can you please tell me where can i get some sip massage text file
to practice translatorX
thanks
Hi Dixon,
Make some SIP calls. Trace them. All SIP devices generate these messages. If you do not have access to SIP devices, there are many websites that break down SIP messages.
Here is a great one:
www.3cx.com/blog/voip-howto/sip-messages/
-John
@@SunsetlearningInstitute
Thank you
good tutorial sir
Glad you liked it!
very usefel thank you very much...
Thanks a lot
how to setting the cause 21 (call rejected)
I really like this video. The volume is a bit low but overall great tutorial!
Thank you for the feedback. We are glad you found it helpful! In the future, we will work on better sound quality :)
Please share this ppt.. I need it pls
Hello, we're glad you found this helpfull! We have several resources like this within our neXT Learning Community portal. It also includes a discussion board to chat with our instructors. If you are intersted, you can email neXTLive365@sunsetlearning.com and we will get you access into the community for free!
Hi sir how to use virtual cube in my laptop that can integrate with cucm.
Hi Prashant,
I am not exactly sure which product you’re asking about. There are two that come to mind:
Cisco has a vCube - a virtual CUBE that runs inside of their UCS/VMWare servers. This is a something you would use in a production environment, instead of using an entire physical router.
The other product you might mean is GNS3. This is used for self-study and practicing learning the commands. - someone who doesn’t not need a real production CUBE. GNS3 is software that you can run on our laptop, it allows you to install and work with a real router operating system (IOS) assuming you have one available to use.
In my video, I was using a physical 2911 router/CUBE via telnet.
JOHN MEERSMA
@@SunsetlearningInstitute pls share reaming videos of SIP
@@krk1848 Thanks for the suggestion!