Senior VoIP Support Engineer (L2L3) – SIP, RTP, Softswitch/SBC & Linux - Onsite Ahmedabad - UK Overlap Hours

Talent Hunters

Ahmedabad District

Sur place

INR 1 200 000 - 2 400 000

Plein temps

Il y a 29 heures
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Résumé du poste

Talent Hunters Oy is seeking a Senior VoIP Support Engineer (L2/L3) based in Ahmedabad to ensure the stability and quality of real-time communication platforms. You will be the escalation point for complex SIP, media and platform issues, collaborating with customers, NOC and engineering teams to diagnose, restore service and prevent repeats.

This hands-on role focuses on troubleshooting, incident management, and proactive monitoring, with opportunities to mentor junior staff and contribute to

Qualifications

  • 4–7 years in VoIP/telecom support or operations.
  • Strong understanding of SIP, SDP, RTP/RTCP and call flows.
  • Hands-on experience with Kamailio, FreeSWITCH or Asterisk.
  • Proficiency with packet analysis tools (Wireshark, sngrep, tcpdump).
  • Solid Linux administration skills (CentOS/Debian/Ubuntu).
  • Understanding of NAT, firewalls, TCP/UDP, DNS, and VPN/IPsec in VoIP.
  • Knowledge of MySQL/PostgreSQL and basic querying for CDR analysis.
  • Experience with monitoring/ticketing tools (Zabbix, Grafana, Prometheus, Nagios, Jira, Zendesk).
  • Strong English communication, written and verbal.

Responsabilités

  • L2/L3 escalation point for VoIP incidents, including call failures and poor call quality.
  • Troubleshoot SIP signalling and RTP media using captures and traces.
  • Support and maintain softswitch, SBC, and proxy platforms (Kamailio/OpenSIPS/FreeSWITCH/Asterisk).
  • Monitor platform health and call quality (ASR, MOS) and act on alerts.
  • Work with carriers to resolve routing, codec, and signalling issues.
  • Own incidents through closure with customer updates and RCA reports.
  • Handle trunking, dial plans, number routing, codecs, NAT traversal and security policies.
  • Implement security practices (fraud detection, rate limiting, ACLs, TLS/SRTP).
  • Write runbooks and SOPs; mentor junior engineers; provide feedback to engineering.

Connaissances

VoIP/telecom experience
SIP, SDP, RTP/RTCP
Kamailio/FreeSWITCH/Asterisk
Packet analysis tools (Wireshark, sngr

Outils

Kamailio
FreeSWITCH
Asterisk
Wireshark
sngrep
tcpdump
Linux (CentOS/Debian/Ubuntu)
MySQL/PostgreSQL
Zabbix
Grafana
Prometheus
Nagios
Jira
Zendesk

Description du poste

We have a rare opportunity for an experienced Senior VoIP Support Engineer (L2/L3) based in Ahmedabad to join one of the UK’s most awarded and fastest-growing next-gen VoIP providers, delivering high-quality digital phone systems and broadband with 5-star service. Positioned as the UK’s No.1 VoIP provider ahead of the 2027 BT Landline Shutdown, this is a fast-paced, award-winning environment with strong opportunities to learn, grow and build your career.

This is a hands-on support and operations role. It does not include software development, DevOps or SRE responsibilities.

As the Senior VoIP Support Engineer, you’ll own the stability and quality of the company’s customers’ real-time communication platforms. You’ll be the escalation point (L2/L3) for complex SIP, media and platform issues, working with customers, NOC and engineering teams to diagnose problems quickly, restore service and prevent repeat incidents.

Must Have Skills and Experience (These are minimum requirements, not nice to haves)
  • 4-7 years in VoIP/telecom support or operations.
  • Strong understanding of SIP, SDP, RTP/RTCP, and related protocols and call flows.
  • Hands-on experience with at least one of Kamailio, FreeSWITCH, or Asterisk.
  • Proficiency with packet analysis tools (Wireshark, sngrep, tcpdump).
  • Solid Linux administration skills (CentOS/Debian/Ubuntu), including shell, logs, networking, and services.
  • Understanding of NAT, firewalls, TCP/UDP, DNS, and VPN/IPsec in VoIP environments.
  • Working knowledge of MySQL/PostgreSQL and basic querying for CDR analysis.
  • Experience with monitoring and ticketing tools (Zabbix, Grafana, Prometheus, Nagios, Jira, or Zendesk).
  • Strong English, both written and verbal communication, with the ability to handle customer escalations calmly.
Key Responsibilities
  • Act as L2/L3 escalation point for VoIP incidents, including call failures, one-way audio, registration problems, poor call quality, and routing issues.
  • Troubleshoot SIP signalling and RTP media using packet captures and call tracing (Wireshark, sngrep, tcpdump, HOMER).
  • Support and maintain softswitch, SBC, and proxy platforms (Kamailio, OpenSIPS, FreeSWITCH, Asterisk, or similar).
  • Monitor platform health and call quality (ASR, ACD, PDD, jitter, packet loss, MOS) and act on alerts proactively.
  • Work with carriers and interconnect partners to resolve routing, codec, and signalling issues.
  • Own incidents through to closure, including clear customer updates, RCA reports, and corrective action plans.
  • Handle configuration work: trunks, dial plans, number routing, codecs, NAT traversal, and security policies.
  • Apply security practices such as fraud detection, rate limiting, ACLs, and TLS/SRTP, and help mitigate toll fraud and SIP attacks.
  • Write and maintain runbooks, knowledge-base articles, and SOPs.
  • Mentor junior support engineers and review their escalations.
  • Give structured feedback to engineering on recurring defects and improvements.
  • Available to attend periodic maintenance windows during off-peak hours.

Due to high application volumes, only shortlisted candidates will be contacted - but we genuinely review, appreciate and value every application and wish you the very best in your search. Please follow us for future roles.

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