The Ultimate SIP Tutorial
HTML-код
- Опубликовано: 30 сен 2024
- This video is a review of a SIP trace using wireshark. The actual call scenario is a call transfer from a phone inside the session border controller to a phone on the outside of a session border controller. This video is intended for people wanting to learn more about SIP.
That pleasant feeling when someone offers you a free help and still says "Thank you for your attention" :)
Great content. Thank you.
Not skipping the adverts to make sure you get at least a little as a thank you from viewers.
Thanks for posting these videos! The company I work for does not provide great training material and this has been helpful to learn the sip process beyond looking up what error codes mean.
Great, glad it is helpful!
Thank you for making this video. It’s helping me a lot to get Spectrum ESG to work with UniFi talk.
Hey, did you ever resolve this? I had a similar problem recently and it was an Ack not being sent and the call cut off after 20 secs. When I added some config to the SBC to forward and fix the ACK to the itsp the call stayed up and the problem was resolved.
Hey Terrell, did you ever find out what the problem was?
Hi such a nice work, how did you get traces from different machine and combine them into one file?
Great tutorial. Thank you! I struggle to follow call ladders on a daily basis... no more.
Did you end up isolating the issue? Haven’t finished the video but it looks like it’s doesn’t trust the remote IP phone and disallows direct media. So was the the solution anchoring media to SBC?
Great video. The tracing of the call flow provides a perspective of how much is happening in such a short period of time. Thank you for your hard work.
Hiw do i get intouch, i need your support service.
Thank you ever so much for taking your time out to educate others, was wonderful to get a basic understanding on SIP tracing :)
Dude, thanks for the tutorial, will it be a problem for you to upload the .pcap files somewhere? It will be great to have those as reference.
Let me see if I can find them. If not, i will try to make some new videos and include the pcaps.
If you click to Media Port: by RightMouseButton and select "Apply as Column" then rewrite that SDP Media Port will be easier.
Very good toturial, I hope you are well and you publish some new video, maybe with FreePBX Toturial ...
Yay I can understand your video
Terrell thank you as this was very helpful.
Thank you for these videos sir!!! Excellent explanation of SIP!
Nice but How could I build my own SIP trunk provider? Thanks
thanks for the content, really interesting !
How is it possible to regroup different traces with wireshark like you do during the video ?
So if all of the connections are using different port numbers... what is using the 5060 port numbers that you see on the wireshark diagram? Is that for the signaling of the INVTES, REGISTERS, 180 Ringings, and PRACKS themselves? And it's the actual data/media that uses the port numbers like the 6380, and 20104?
Also, how is the phone making all of these invites and transfers? This may be a silly question, but are all of the invites, from 192.168. 230.102 all from literally the same physical phone?
And lastly... are RTP / SRTP packets sent when no one is speaking? How does the phone know what constitutes 'silence'? perhaps there's some faint background noise, but the person isn't actively speaking, will it still drop the call after a while? If the person remains quiet on the line, Will is still send RTP packets of voice data, but just be silence?
Enjoyed very much, hope you get feeling better.
Just a small correction around @44:20, the odd number port are RTCP. Great video really enjoyable
it worth it subscribing to your channel , thanks a lot
thank you for this nice video on SIP.
Great breakdown, thanks so much.
Video was nice but this is not for beginners at all. Too disorganized and whipping out wireshark first thing is not helpful to a beginner.
Thanks for sharing and certainly a lot of good fault finding pointers.
Hi Terrell, I'm watching your vid but it is still a bit advanced for me. As a Avaya eng. I want to be proficient in SIP and SBCs. There are so many things to learn. Where should I start?
i'd recommend sip school or reading the RFC, also playing around with wireshark (or snooper if you are using lync) using a simple setup(2 phones on the same LAN for example), and making searches on google for things you dont understand. I will take time to understand but you'll gain experience. there are a lot of webinar videos on youtube talking about troubleshooting voip using sip traces..
👍
hello Terrell,
Great great video! I am very new to VoIP and got a little bit confused... Could you please clarify the following:
Based on the wireshark capture I can see that the caller (leftmost phone) requests on-hold and then redirect. Why would it do that? I though that on-hold and such redirect should have come from the ITSP. Am I missing something obvious?
The Music On Hold comes from the PBX itself. In order for that to play, the audio path must be connected to the PBX.
You rock thanks bro
Thank You!
excellent video, still relevant and helpful today.
Thank you
excellent scenario choosen ,it's very helpful man
Really informative.helps develop an understanding on how to see wiresharks and troubleshoot call related issues.You should do some more of these which will help us to actually see the trace information elements headers in a trace,how the flow should be and behavior of network nodes.I only suggest in order to save time you can create the diagrams before hand and can show us call instances to it.
I am trying to set up my wireshark 2.2.2 to read the particular packet once I click on flow stream but it is not happening as you have shown.can you tell us which wireshark version are you using?
Keep up the great work.wishing to see more of such troubleshooting videos. :)
This is great feedback, Thank you. I agree about setting up the diagram in advance. I am using version 2.2.1. Make sure you don't have any filters applied to your trace when trying to click on the flow stream.
Very informative..Great work Terrell..
G711 is what is recommended to use on the WAN side of the SBC for bandwidth saving.. Unless the ITSP requires something else..
G711 is not compressed at all, so I would not consider that a bandwidth saver. I would use G723, or G729. Check out this table to understand bandwidth consumption by codec. www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html
@@TerrellBoyer Should we not used first g723, g729 and in the end g711? as I have also same codec g711U, g723, g729 , but I think if I change priorities will it be more economical in bandwidth? and what about Opus?
thanks for share
Awesome. Thank you.
Thank very much :) :) :)
This is great ! thanks ...
omg... your voice!