SIP Troubleshooting for Beginners - Outgoing Call Trace Review

[@penguin7776]

I start VOIP training in a couple of weeks. 8 hours all week. I have the criteria and want to get a jump on it. One was reading SIP Traces. This video was pretty easy to grasp. Thanks.

[@TerrellBoyer]

Thanks for the comment.

[@TerrellBoyer]

Where do you get voip training?

[@penguin7776]

I work at GTT. I do internet troubleshooting and recently acquired VOIP. We have a couple of VOIP engineers flying up from Dallas to train us. 8 hour for 5 days next week =fried brain.

[@TerrellBoyer]

+Rich Yanick Very Cool...

[@samuelsmith7548]

Going for a CO field position with Verizon... Thanks for the overview... Will help with my interview!

[@949surferdude]

Thank you very much. Out of all the SIP videos on KZbin yours was the most effective in explaining the SIP call flow. Can you do a video on building a SIP trunk. I don't understand the basics needed to build one (coming from a Avaya perspective)

[@williamburling3229]

I would appreciate your giving a presentation on how to set up wireshark or some other free app to enable us to see what you are seeing. Thank you for taking your valuable time to help us

[@ericblair9756]

Email me [e-mail] I've got ya

[@MarieColaco]

Thank you so much Terrell Boyer. You explain the SIP message in a very simple & precise way.

[@ECrespo175]

Great video, I would like to see more videos about deciphering each message (100,183,200) in detail.

[@TerrellBoyer]

Thanks Ed, I will work on that.

[@cjamesmusic]

Very informational. I needed to brush up on basics for a voip tech job interview and this was one of the videos i watched. Thanks!

[@jakecormier3827]

Great instructions and explanations. Still applies today in 2019

[@ankitdhyani961]

Great video, Terrell. Detailed and to-the-point explanation.

[@ADJ161996]

Man, this is very helpful. Thank you very much for the informative SIP content

[@willbuck7952]

Outstanding job-sharing your knowledge is commendable. I salute you sir.

[@privera0933]

Nice video. It would be greater if you could do a video on how to troubleshoot Jitter issues. Keep them coming!

[@huyentruong1269]

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[@brightorb1]

thanx so much for the amazing post , please please provide more simple to understand sip analysis

[@templedogs7847]

Terrell,, Thank you for making the videos and sharing what you have learned with those that are learning it...Such is the circle of life. Subscribed and looking forward to more from you.

[@flower789ash]

This video tutorial is awesome ,Please do a tutorial with SIP PRACK Call flow

[@rythmiccool]

Nice Video Terrell with a basic understanding of call flow.

[@williamcastro4171]

Excellent video. Thank you for your time and dedication to share your knowledge!

[@michaelfrederickong7519]

Nicely done , I am just starting in this kind of job and you made so easier to understand.

[@gesusdube]

Excellent video...you have explained things in much better simpler way than my teachers !!! I have a question to ask you-I have a site using SIP trunks. When I dial a 4 digit extension from a shared line, it gives my fast busy tone ONLY in SRST mode If I dial the same number from same phone in normal mode, all works well. My suspicion is Cisco Toll Fraud Prevention is blocking my calls...maybe,,,, For this, I have captured traffic using wireshark (when I unable to dial the number) but I don't know where in wireshark in will give an idea that it is indeed toll fraud prevention blocking calls!!!I ave been looking and looking. Any suggestions?

[@TerrellBoyer]

I would look for your specific call in the trace and look to see why it was rejected. If Cisco uses SIP protocol for their stations, you should at least see the call being initiated. Once you see that, follow the SIP flow to see what the rejection code was. It may not tell you specifically, but it may give you a better idea.

[@graham8377]

Thanks! Good video. I really liked that filter tip to see only the VoIP call.

[@masterofkings7887]

Hi Terrell, this is very helpful. I am supposed to demonstrate the same to my colleagues in a training session so that I can easily explain about encrypted SIP in Microsoft Lync calls. I searched for a completed SIP call capture file in wireshark site, but couldn't find one that is as good as this. I would be glad if you could share this capture file with me. Thank you.

[@vindasad]

Wow, this is definitely the video I was looking for, really good explanation, Terrel I do not know how to express how grateful I am with this video, I hope you can make more videos such as this one... Thanks :) Subscribed!!!!

[@vickneswaran8506]

Thank you - always help to clear up your understanding.

[@benice3117]

Can you please show how and where you setup the trace in relation to where your equipment and firewall is. If you could maybe throw in a diagram that would be great. I'm confused where you took the trace and what port was setup for mirroring and such.

[@petermuia9519]

Hi Terrell. This is a good video. I was wondering like Crusty Tackleford,(who asked 10 months ago) how you setup your equipment to be able to capture these packets with Wireshark. Please shade light on this

[@s.m.ehsanulamin7235]

For making an external call from my PBX to outside by means of sip trunk , i have experienced an problem. The calls are diconnected after one mins. The sip trunk were configured in SBC. I donot know what should i do? I checked the wireshark trace and found that the Release Message were coming from the Phone . Internal calls are woking fine. Will be glad to have your suggestion. Thanks in advance.

[@magpieenterprise6781]

When would you take a packet capture and when to take call logs?

[@rajendranalawade3239]

great post Terrel, it gives good understanding of call flow.

[@pranabpadhi]

Thanks for sharing Terrell, keep up the good work.

[@bshack0]

Thanks for the video! Greatly helped me understand call tracing at the SIP level.

[@The-practice]

Hello, not sure if you still respond here. Just trying. I watched your outbound tutorial on SIP Troubleshooting for Beginners - Outgoing Call Trace Review. I am needing to understand how to set up traces to show the RTP stack. Currently I am unable to figure out how to incorporate the RTP/Audio information in my PCAPS. Any help on this would be appreciated. Thanks, Raheem

[@ralshwk]

Great video and explanation. Please make more. Thanks

[@ernestoserrano946]

how would I answer this questions? You have observed the INVITE - 200 OK - ACK three-way handshake during the call setup. What messages are exchanged for tearing down a call session?

[@ryanmcmillan763]

Thanks for the video, very simple and easy to follow.

[@hopefortruth]

Perfect! Thanks for sharing. I will be checking in for more videos!

[@oussverde]

thanks for sharing very helpful and clarifying

[@kodangbryan9662]

Great work pls i need help on SIP congestion

[@conradbennett6961]

Thanks bro for your work, its really a great intro to an SIP...

[@musememedia3429]

This was a great video. Thanks for posting this stuff, its really valuable!!!

[@ramrathods]

Great Video Terrell. Hats off!

[@TerrellBoyer]

Ram Rathod Thank you!

[@ankitmunhet4659]

thank you Terell you have given me a great information which i was looking for

[@Lordvishnus]

Thank you Terell..Do you have any real time examples for choppy audio, one way audio , audio gaps in SIP protocols..

[@lekepope]

Terrell, you the man!!!!Please keep it up

[@shaiz1985]

from where can i get the sip training in Riyadh Saudi Arabia, a physical taring as i am working in STC and can understand the system, traces, putty etc , please your feedback

[@alltech247]

Hi Terrell THUMBS UP....This is great tutorial.

[@kingshuksinha3061]

Nice explanation. Hope to see more from you..Awesome

[@peyton05220]

Lots of information, this makes me learn!

[@TerrellBoyer]

Thanks Son!

[@Ayelmani]

Great video, very helpful. Thank you. How to capture whether DTMF is in band or out of band?

[@onyxsolo1]

Hi Ayman, It will be in the messaging and is determined during the call setup based on my understanding. All media gateways should detect DTMF tones, they let the PBX/CFS/Switch or whatever you're using know it detected tones and what the digits were so the PBX etc can determine if they have a feature associated with it or not (flashook/3way calling etc). If it doesn't those digits get passed over to the endpoint device of the sip trunk the call routed over; However, as I stated earlier whether the DTMF tones are passed inband or out-of-band is determined when the call is first setup based on my experience so you should see it in your standard sip messaging capture.

[@mozbius]

Can you do the same type of video for a call transfer?

[@romanislam1805]

Hi Terrell, Do you teach VoIP online ? or do you know any good training institute ?

[@BrianThomas]

Great video Terrell. I'd love to setup an ADTRAN 908e in a lab in order to capture a pcap file with a good working test calls. Thanks to your video's I think I have handle on setting up for the tcpdump. I'm just not sure how to setup an ADTRAN 908e for a lab environment. Do you have any suggestions? I've contacted ATRAN, but no luck yet. The ADTRAN does have some great debug commands that I've used many times, but not as good as what you'll get in Wireshark.

[@TerrellBoyer]

What are you looking to setup?

[@BrianThomas]

Terrell Boyer I'm looking to setup a PRI lab using the 908e

[@TerrellBoyer]

Well, If I gain access to a 908e, I will keep that in mind for a future video.

[@FahadullahMuhammad]

How did the call last for 14 seconds, when the start time is 10 and stop time is 14? Please explain.

[@stevenfrazier7959]

Great job, thanks very much!

[@santoshr351]

Thanks for this video. This is very useful. Keep up the good work!!

[@ccie8340]

Terrell, Excellent Video..Simple and to the point. Thanks for sharing this.

[@renjithknair7724]

hello sir what does it mean 401 Unauthorized and 500 Internal server error . SIP outgoing call not working after analyses the packet flow i received this

[@arizshakilkhan6039]

Thanks for making things easier!!

[@sebastianolvianboros2083]

Hello, @Terrell. Great Video btw. Is there any chance to receive RTP packages ( from freeshwitch) while outgoing / incoming calls only (no active session)

[@grmetechnologies3967]

Thanks for the video. Learnt a lot.

[@nestorguzman5018]

Great explanation! Thank you Terrell.

[@jagan1991]

its really great... expecting many more ....

[@TerrellBoyer]

Jaga Priyan Thanks for the comment. Planning to make more SIP tutorials this week!

[@wjoybrown]

Very helpful... thank you sharing your knowledge.

[@PawanSharma-jn6um]

Can we convert syslogs to pcap file to view the call flow in wireshark?

[@namle-br8ju]

Thank a bunch for sharing a great video

[@jwebb1975]

This is a great video. Nice work.

[@ajith_k]

Awesome, simple and very informative. Thanks for this :)

[@zdye14]

Great tutorial!!! Thanks for sharing your knowledge ..

[@mikemiller5051]

Great video!!

[@ccntwc]

Awesome video! thank you.

[@satheeshkumar-it7pz]

Excellent video, Thank Bro!!

[@bigwizzle45]

easily digestible info. Thanks

[@TomJerrysVlogs]

Well explained.

[@kkupadhayay]

its really great terrell......

[@cutesammie]

Great Video. Thanks a lot.

[@andyf424]

Great tutorials!

[@godin7312]

Very good, informations important, is a help enough

[@johanngonzalez9780]

Good Video !!! congrats

[@NBAUsaisyourfather]

Seriously lovee it

[@stephaniem3149]

your voice is just like a chocolate's commercial :p

[@thebuckstopshere79]

great video - but i think the call length was around 4 sec - not 14

[@TerrellBoyer]

thebuckstopshere you are orrect. Good catch.

[@thebuckstopshere79]

Terrell Boyer Still an excellent video. Probably one of the best SIP call flow tutorials on the Web. Congrats

[@joelourenco4621]

Thank you!

[@logandrake4946]

thank you a lot that was so helpful

[@MrVenugopal2010]

Good one

[@thompsonlin7249]

thank you for the vedio

[@Eskimoz]

Top !

[@xiaomeng3943]

I could be more helpful with the topology diagram, which shows how everything is connected.

[@sultanfahad]

Very useful and helpful. Thanks!

[@elhabibbirouk9722]

Thank you!