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Sip Rtp Developer Jobs (NOW HIRING)

Zenith System Solutions Inc

... Handle SIP signaling, RTP/SRTP, SDP, DIAMETER, transcoding, and telecom routing • Lead ... Bachelor's/Master's degree in Electrical, Electronics, Telecommunications, or Software Engineering ...

Inabia Software & Consulting Inc.

IMS Engineer

Ridgeland, MS · On-site

$68K - $91K/yr

Bachelor's degree in telecommunications, computer science, engineering, or a related field; advance ... Strong technical expertise in IMS protocols, including SIP, RTP, Diameter, HSS and related ...

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How much do sip rtp developer jobs pay per year?

As of Jun 7, 2026, the average yearly pay for sip rtp developer in the United States is $129,348.00, according to ZipRecruiter salary data. Most workers in this role earn between $106,000.00 and $157,000.00 per year, depending on experience, location, and employer.

What are some common challenges faced by SIP RTP Developers when integrating VoIP solutions with existing communication systems?

SIP RTP Developers often encounter challenges related to interoperability when integrating VoIP solutions with legacy telephony infrastructure or third-party platforms. Ensuring seamless audio quality, managing NAT traversal, and handling codec negotiations are frequent technical hurdles. Additionally, developers must frequently troubleshoot signaling issues and maintain compliance with evolving SIP and RTP standards. Collaborative work with network engineers and QA teams is essential to address these challenges and deliver robust, scalable communication solutions.

What are the key skills and qualifications needed to thrive as a SIP RTP Developer, and why are they important?

To thrive as a SIP RTP Developer, you need strong expertise in VoIP protocols (SIP, RTP), network programming, and a background in computer science or a related field. Familiarity with tools like Wireshark, Asterisk, FreeSWITCH, and experience with SIP stack libraries and real-time communications frameworks are typically required. Strong problem-solving abilities, attention to detail, and effective communication skills help developers address complex integration and troubleshooting challenges. These skills are crucial for ensuring reliable, secure, and high-performance real-time communication solutions in modern telephony systems.

What is the difference between Sip Rtp Developer vs VoIP Network Engineer?

AspectSip Rtp DeveloperVoIP Network Engineer
CredentialsExperience with SIP, RTP, VoIP protocols, and programming skillsNetworking certifications (e.g., CCNA, CCNP), knowledge of VoIP protocols
Work EnvironmentSoftware development focused, often in telecom or VoIP companiesNetwork infrastructure, troubleshooting, and deployment in enterprise or service provider settings
Industry UsagePrimarily in VoIP application development and integrationNetwork design, maintenance, and optimization for VoIP services

While both roles involve VoIP technologies, Sip Rtp Developers focus on coding and developing SIP and RTP-based applications, whereas VoIP Network Engineers handle the deployment, configuration, and troubleshooting of VoIP networks. The roles often overlap but serve different aspects of VoIP solutions.

What are SIP RTP Developers?

SIP RTP Developers are software engineers who specialize in developing and maintaining applications that use the Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP) for voice, video, and messaging communications over IP networks. They work on VoIP (Voice over Internet Protocol) solutions, integrating call control, signaling, and media streaming functionalities. Their responsibilities include designing protocols, troubleshooting call quality issues, and ensuring systems are secure and scalable.
Infographic showing various Sip Rtp Developer job openings in the United States as of May 2026, with employment types broken down into 50% Full Time, and 50% Contract. Highlights an 50% In-person, and 50% Remote job distribution, with an average salary of $129,348 per year, or $62.2 per hour.
Senior Software Engineer with SIP/RTP & VoIP exp. At Toronto, Canada/ Milpitas, CA

Senior Software Engineer with SIP/RTP & VoIP exp. At Toronto, Canada/ Milpitas, CA

Aptiva Corp

Milpitas, CA • On-site

$142K - $188K/yr

Contractor

Posted 4 days ago

Job description

 
 

Senior Software Engineer – C++, SIP/RTP Expert

Location:  Toronto, Canada/ Milpitas, CA

Department: Software Engineering

Client is a leader in lawful intelligence solutions providing real-time data insights for telecommunications and cybersecurity domains. Our engineers work on cutting-edge communication interception and media analysis technologies, including GVMC, Dialogic XMS, SIP, RTP, and VoIP call processing.

Job Overview:

We are seeking a highly skilled Senior Engineer with expertise in voice media controllers (eg. GVMC), Dialogic XMS, and deep knowledge of SIP/RTP protocols. The ideal candidate will have a strong background in C++ and Java development, with experience in packet analysis (Wireshark/PCAPs), debugging SIP/RTP issues, and designing scalable communication solutions.

Required Skills & Qualifications:

•             Expert knowledge of SIP, RTP, and VoIP protocols (Call Setup, Media Negotiation, SDP, ICE, STUN, TURN).

•             Deep understanding of media processing frameworks such as GVMC, Dialogic XMS, FreeSWITCH, and Asterisk.

•             Strong experience in C++ and Java development for real-time communication applications.

•             Hands-on experience analyzing SIP, RTP, and T.38 FAX PCAPs using Wireshark/tcpdump.

•             Experience with SIP signaling flows, error handling (503, 408, 487, etc.), and debugging call failures.

•             Proficiency in media codecs (G.711, G.722, AMR-WB, Opus, H.264, VP8, MPEG).

•             Experience in deploying GVMC/Dialogic XMS in Kubernetes (K8s) clusters.

•             Knowledge of distributed logging and monitoring tools (ELK Stack, Prometheus, Grafana).

Preferred Skills

•             Experience with WebRTC, IMS, and VoLTE/ViLTE protocols.

•             Knowledge of network security (TLS, DTLS, SRTP) for VoIP.

•             Experience with cloud-native architectures (GCP, AWS, Azure).

•             Familiarity with CI/CD pipelines (Jenkins, GitHub Actions) for VoIP applications.

•             Hands-on experience in high-performance media gateway solutions.

Education & Experience

•             Bachelor’s or Master’s degree in Computer Science, Electrical Engineering, or related field.

•             8+ years of experience in VoIP, SIP, RTP, and media processing solutions.

•             Prior experience working with GVMC, Dialogic XMS, or similar media servers.

Key Responsibilities

•             Develop and enhance media processing solutions using GVMC and Dialogic XMS using MSML APIs.

•             Implement, troubleshoot, and optimize SIP and RTP call flows for VoIP and video communication.

•             Analyze and debug packet captures (PCAPs) using Wireshark, tcpdump, and other network debugging tools.

•             Design and optimize real-time media streaming solutions for VoLTE, ViLTE, and multi-party video conferencing.

•             Develop and maintain C++ and Java-based software for VoIP and media processing applications.

•             Integrate, test, and validate SIP-based call sessions, ensuring protocol compliance with ATIS 0700005, T1.678 V2, ETSI 102-232 Part 5 & 7.

•             Collaborate with QA and DevOps teams to automate testing for SIP, RTP, and media services.

•             Optimize transcoding and media handling for H.264, H.263, VP8, and MPEG video streams.

•             Contribute to architectural design decisions, ensuring system scalability and high availability.

•             Work with Kubernetes-based deployments for GVMC/Dialogic XMS in cloud environments (GCP, AWS, Azure).