Roadmap SIP Flow Troubleshooting Practical Lab Interview Questions FAQ

30-Day VoIP Fresher Roadmap: Learn SIP, RTP, Linux & VoIP Troubleshooting

A practical, interview-focused learning path for beginners targeting VoIP, SIP, Telecom Support, NOC or Junior VoIP Engineer roles. Learn networking fundamentals, SIP signaling, RTP, SDP, codecs, Linux, Wireshark, FreeSWITCH and Asterisk, followed by practical troubleshooting exercises and 100 VoIP interview questions.

NetworkingSIPRTP / SDP LinuxWiresharkFreeSWITCH / Asterisk

What the fresher should be able to do

Target outcome: Understand a SIP call, explain the signaling flow, inspect SDP/RTP in Wireshark, troubleshoot basic call failures and audio issues, work comfortably on Linux, and explain the role of Asterisk/FreeSWITCH.

Skill Priority

SkillPriorityInterview Importance
Networking★★★★★Must know
SIP★★★★★Core VoIP skill
RTP / SDP★★★★★Core troubleshooting
Linux★★★★★Very important
Wireshark / tcpdump★★★★★Major differentiator
VoIP troubleshooting★★★★★Critical for support roles
Asterisk / FreeSWITCH★★★★☆Strong advantage
Codecs★★★☆☆Basic knowledge
Bash / Python★★★☆☆Useful later
SS7 / advanced telecom★☆☆☆☆Learn later
30-Day Roadmap

30-Day Learning Plan

Days 1–7 · Networking Fundamentals
  • IP, MAC, subnetting, gateway
  • Public vs private IP
  • TCP vs UDP
  • Ports and protocols
  • Routing and static routes
  • NAT, SNAT, DNAT, PAT
  • Firewall and VLAN basics
  • Linux commands: ip, ping, ss, tcpdump
Days 8–14 · SIP Fundamentals
  • What is SIP and VoIP?
  • User Agent, UAC, UAS
  • Proxy, Registrar, B2BUA
  • SIP trunk and gateway
  • INVITE, ACK, BYE, CANCEL
  • REGISTER and OPTIONS
  • 100, 180, 183, 200
  • 401, 403, 404, 408, 480, 486, 487, 488, 503
  • Analyze real SIP traces
Days 15–21 · RTP / SDP / Wireshark
  • Understand SDP fields
  • RTP and RTCP
  • RTP IP and port
  • Codec negotiation
  • G.711 PCMA / PCMU
  • G.729, Opus, AMR basics
  • Jitter, latency, packet loss
  • Wireshark SIP/RTP analysis
  • No-audio and one-way-audio labs
Days 22–24 · Linux Fundamentals
  • Files and directories
  • grep, find, tail, less
  • systemctl and journalctl
  • CPU, memory and disk checks
  • ip addr and ip route
  • ss -lntup
  • Process and service troubleshooting
Days 25–26 · Asterisk & FreeSWITCH
  • Asterisk fundamentals
  • PJSIP, endpoints and trunks
  • Dialplan and contexts
  • FreeSWITCH fundamentals
  • Sofia and SIP profiles
  • Gateway, dialplan and bridge
  • Basic CLI troubleshooting
Days 27–30 · Interview & Practical Preparation
  • Build an end-to-end SIP lab
  • Capture SIP and RTP
  • Break and repair the lab
  • Practice troubleshooting scenarios
  • 100 interview questions
  • Mock interview
  • Final practical assessment

Core SIP Call Flow

Caller Callee | | |-------- INVITE ------------>| |<------- 100 Trying ----------| |<------- 180 Ringing ---------| |<------- 200 OK --------------| |-------- ACK ---------------->| | | | RTP MEDIA | |<============================>| | | |-------- BYE ---------------->| |<------- 200 OK --------------|

The Most Important Troubleshooting Model

SIGNALING

Call doesn't reach server

Check interface → IP → route → firewall → port → packet capture.

SDP

Call connects but no audio

Check negotiated media IP, RTP port, codec and SDP direction.

RTP

One-way audio

Verify RTP in both directions. Check NAT, firewall, routing and advertised media address.

Essential Linux Commands

ip addr
ip route
ip route get 8.8.8.8
ss -lntup
ping 8.8.8.8
traceroute 8.8.8.8
tcpdump -i any -nn port 5060
tcpdump -i any -nn host 10.10.10.10
systemctl status <service>
journalctl -u <service>
df -h
du -sh *
free -h
top
ps aux

Wireshark Filters

sip
rtp
udp
ip.addr == 10.10.10.10
sip.Method == "INVITE"
sip.Status-Code == 200

Recommended Practical Project

VoIP SIP Troubleshooting Lab

Linux Server | FreeSWITCH | ┌────────┴────────┐ │ │ Softphone A Softphone B

Demonstrate: SIP registration, authenticated REGISTER, SIP call flow, SDP negotiation, RTP packets, codec negotiation and Wireshark analysis.

Intentionally break: SIP port, RTP port, firewall, route, codec and media path. Document how each problem was identified and fixed.

100 Interview Questions

Networking & SIP Basics (1–20)
  1. What is an IP address?
  2. What is a MAC address?
  3. What is a subnet mask?
  4. What is a gateway?
  5. What is DNS?
  6. What is DHCP?
  7. What is ARP?
  8. What is NAT?
  9. What are SNAT and DNAT?
  10. What is PAT?
  11. What is a VLAN?
  12. TCP vs UDP?
  13. What is a port?
  14. What is a routing table?
  15. What is a default route?
  16. What is ICMP?
  17. What does ping do?
  18. What does traceroute do?
  19. How do you check Linux IP configuration?
  20. What is VoIP?
SIP, RTP & Media (21–50)
  1. What is SIP?
  2. What is a SIP User Agent?
  3. What are UAC and UAS?
  4. What is a SIP proxy?
  5. What is a registrar?
  6. What is a SIP trunk?
  7. What is a SIP gateway?
  8. What is an SBC?
  9. What is INVITE?
  10. What is ACK?
  11. What is BYE?
  12. What is CANCEL?
  13. What is REGISTER?
  14. What is OPTIONS?
  15. What is REFER?
  16. What is 100 Trying?
  17. What is 180 Ringing?
  18. What is 183 Session Progress?
  19. What is 200 OK?
  20. What is 401?
  21. What is 403?
  22. What is 404?
  23. What is 408?
  24. What is 480?
  25. What is 486?
  26. What is 487?
  27. What is 488?
  28. What is 503?
  29. What is RTP?
  30. Why is RTP usually UDP?
Troubleshooting & Linux / Platforms (51–100)
  1. What is RTCP?
  2. What is SDP?
  3. What is a codec?
  4. What is G.711?
  5. PCMA vs PCMU?
  6. What is G.729?
  7. What is Opus?
  8. What is AMR?
  9. What is codec negotiation?
  10. What is transcoding?
  11. What is an RTP port?
  12. What is payload type?
  13. What is an RTP sequence number?
  14. What is jitter?
  15. What is packet loss?
  16. What is latency?
  17. What causes poor voice quality?
  18. What causes one-way audio?
  19. INVITE isn't reaching the server — what do you check?
  20. SIP reaches server but no response — what do you check?
  21. Why use tcpdump?
  22. How do you capture SIP traffic?
  23. How do you check port 5060?
  24. Call returns 404 — what do you check?
  25. Call returns 403 — what do you check?
  26. Call returns 503 — what do you check?
  27. Call returns 408 — what do you check?
  28. No audio — what do you check?
  29. One-way audio — what do you check?
  30. Distorted audio — what do you check?
  31. Calls disconnect after 30 seconds — why?
  32. Registration fails — what do you check?
  33. Internal calls work but external calls fail — why?
  34. Outbound works but inbound doesn't — what do you check?
  35. Inbound works but outbound doesn't — what do you check?
  36. How do you identify RTP in Wireshark?
  37. How do you identify codec negotiation?
  38. How do you determine where a SIP call failed?
  39. How do you check CPU?
  40. How do you check memory?
  41. How do you check disk space?
  42. How do you check listening ports?
  43. How do you check routes?
  44. What is Asterisk?
  45. What is FreeSWITCH?
  46. What is a FreeSWITCH Sofia profile?
  47. What is a dialplan?
  48. How would you troubleshoot a complete failed VoIP call?

Frequently Asked Questions About VoIP for Freshers

What skills does a VoIP fresher need?

A VoIP fresher should understand computer networking, SIP signaling, RTP, SDP, Linux, Wireshark and basic VoIP troubleshooting. Familiarity with Asterisk or FreeSWITCH is a strong advantage.

Is SIP difficult to learn?

SIP becomes much easier when learned through real call flows. Start with INVITE, 100 Trying, 180 Ringing, 200 OK, ACK, RTP and BYE before moving into advanced SIP features.

What is the difference between SIP and RTP?

SIP is primarily used for call signaling and session control, while RTP carries the actual real-time audio or video media.

How do I troubleshoot a VoIP call?

Start by checking whether SIP reaches the server, then inspect the SIP response, routing, firewall and configuration. If the call connects but has audio problems, inspect SDP and RTP in Wireshark.

What are common SIP interview questions?

Common questions cover SIP call flow, INVITE, REGISTER, ACK, BYE, SIP response codes, SDP, RTP, codecs, NAT, one-way audio, no audio and SIP troubleshooting.

Is FreeSWITCH useful for a VoIP engineer?

Yes. FreeSWITCH provides practical exposure to SIP profiles, gateways, dialplans, codecs, RTP and call routing, making it useful for hands-on VoIP learning.

Is Asterisk required for VoIP jobs?

It is not mandatory for every role, but basic Asterisk knowledge can strengthen a fresher's profile. Understanding one VoIP platform deeply is more useful than memorizing several platforms.

How long does it take to learn VoIP basics?

With consistent daily practice, a beginner can build a solid foundation in networking, SIP, RTP, Linux and troubleshooting within about 30 days. Real lab practice is essential.

Final Interview Checklist

  • Can explain SIP call flow without notes.
  • Can distinguish SIP, SDP and RTP.
  • Can identify common SIP response codes.
  • Can read a basic Wireshark SIP trace.
  • Can identify RTP IP/port and codec from SDP.
  • Can troubleshoot no-audio and one-way-audio scenarios.
  • Can use basic Linux networking commands.
  • Can explain SIP trunk, SBC, gateway and softswitch.
  • Has completed one practical FreeSWITCH/Asterisk lab.
  • Can explain troubleshooting logically instead of guessing commands.
VoIP Fresher Roadmap · Designed for practical interview preparation · GingerBook