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We are looking for an experienced VoIP engineer (OpenSIPS + rtpengine) to debug and fix a NAT-related audio issue in our production SIP platform. We have OpenSIPS 3.3 + rtpengine 1.11 behind NAT. Calls work when SDP contains public IP. No audio when SDP contains private IP. RTP visible on firewall but not on server. Need an expert to debug NAT, firewall, and rtpengine behavior. Must have real OpenSIPS + rtpengine experience. Important: This is NOT a basic SIP config task. We need a senior VoIP engineer who understands deep NAT + RTP edge cases. If you don’t have hands-on OpenSIPS + rtpengine experience, please do not apply.
ID do Projeto: 40170879
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9 freelancers estão ofertando em média $161 USD for esse trabalho

Drawing from my decade-long career as a Network, VoIP, Cybersecurity and System Engineer, I have not only the know-how but also the hands-on experience required to tackle your sophisticated issue. My expertise with OpenSIPS and rtpengine encompasses debugging deep NAT-RTP scenarios like the one you are experiencing. Having worked on similar projects, I understand the urgency and importance of resolving audio issues promptly for uninterrupted communication. Besides my extensive VoIP background, my proficiency with numerous networking technologies/vendors such as Cisco, Mikrotik, Fortinet etc. is another asset that I bring to the table. I design and deploy systems based on industry-best practices which ensure not just quick fixes but reliable long-term solutions. I make myself readily available to communicate, respond swiftly to queries or clarifications, and wholeheartedly commit myself to project completion within timeline. Your unique project requires an expert who can not only fix the immediate problem but also ascertain its root cause - that's where my competency truly shines. Entrusting me with this task will yield results beyond expectation; let's work together and unveil the solution!
$200 USD em 3 dias
5,1
5,1

Hi there, I have 10+ years working experience in VoIP field, worked a lot with OpenSIPS/rtpengine and will be happy to help you with your audio issues. I'll need access to your server (ssh) running OpenSIPS, might need access to your NAT/Firewall device as well (or someone who manages it).
$180 USD em 7 dias
4,0
4,0

Hi, there, I have extensive experience in VoIP engineering, specifically with OpenSIPS and rtpengine. With a deep understanding of NAT and RTP edge cases, I am well-equipped to tackle the audio issue in your SIP platform.✅ I will thoroughly examine the NAT, firewall, and rtpengine behavior to pinpoint and resolve the audio problem.✅ Leveraging my past experience with similar issues, I will ensure that calls work seamlessly even with private IP in the SDP.✅ I will implement robust solutions to make the RTP visible on your server, guaranteeing optimal audio transmission.✅ My hands-on expertise in OpenSIPS and rtpengine qualifies me as the senior VoIP engineer you need to fix this challenging problem.✅ By combining my Telecom and VoIP engineering skills, I will deliver a comprehensive solution tailored to your specific requirements. Looking forward to working with you. Best Regards. Anzhelika
$165 USD em 1 dia
2,2
2,2

Your issue is a classic NAT traversal problem where rtpengine isn't properly rewriting the SDP c= line to replace private IPs with your public-facing address, causing return RTP packets to hit your firewall but never reach the media proxy. I've spent years debugging exactly these OpenSIPS + rtpengine edge cases in production environments. My approach: first verify rtpengine's listening interfaces and `replace-origin`/`replace-session-connection` flags in your `rtpengine_manage()` calls, then trace the full SDP transformation through OpenSIPS to confirm proper media anchoring, and finally validate your iptables/nftables rules aren't dropping legitimate RTP on the designated port range. I can start immediately and typically resolve these NAT/RTP issues within one focused debugging session.
$30 USD em 1 dia
0,0
0,0

Hello Arpit M., We would like to grab this opportunity and will work till you get 100% satisfied with our work. We are an expert team which have many years of experience on Asterisk PBX, VoIP, Telecommunications Engineering, Software Development, FreeSwitch, Telecoms Engineering, Telecom, SIP Lets connect in chat so that We discuss further. Regards
$140 USD em 7 dias
0,0
0,0

If RTP is reaching the firewall but never the media engine, this is a classic NAT traversal and anchoring failure that needs deep VoIP debugging, not guesswork. Hello, I have hands-on production experience with OpenSIPS and rtpengine in NAT-heavy environments, including complex SDP rewriting and RTP anchoring issues. I understand scenarios where calls succeed with public SDP but fail with private IPs, usually involving incorrect rtpengine offer/answer handling, broken direction flags, or NAT misdetection. My approach starts with tracing SIP signaling, SDP bodies, and rtpengine logs together to confirm where media negotiation breaks. I will verify rtpengine kernel module behavior, interface bindings, advertised addresses, and symmetric RTP handling. I will inspect OpenSIPS NAT logic, rtpengine flags, and topology hiding interactions. Firewall state, port ranges, and conntrack behavior will be validated against actual RTP flows. This is not a basic setup task, and I’m comfortable debugging these edge cases live in production. I can identify the root cause quickly and apply a safe, permanent fix.
$250 USD em 7 dias
0,0
0,0

With over a decade of experience in various areas of software development, I have refined my skills such that complex debugging and troubleshooting are an essential part of what I do. In particular, I've spent significant time working with VoIP technologies, including OpenSIPS and rtpengine, which makes me well-equipped to handle your distinct challenge. I understand how even slight tweaks to configurations can have substantial impacts on NAT and RTP which makes me not only suited for this job but also uniquely qualified. Having successfully resolved many deep NAT + RTP edge cases in the past, your problem resonates with my experiences. My ability to analyze logs, review configurations, and understand the interactions between different components allows me to quickly identify the root cause of No Audio issues involving private IPs in SDPs. After that, it's all about implementing targeted solutions that bring your SIP platform back to full functionality. Importantly, I appreciate the complexity and uniqueness of this task. It's not a straightforward matter of basic SIP configuration but necessitates a thorough understanding of intricate network behaviors. Rest assured, your Wisely selected choice saves Time and eliminates Regret; as a full-stack developer known for providing seamless integrations into workflows & teams or spearheading projects driven by technical depth, I am committed to delivering an effective solution promptly. Let's get your audio flowing again!
$200 USD em 1 dia
0,0
0,0

Hi, I have been working in VoIP domain for 20 years. and I have extensive experiences with OpenSIPS and Kamailio. Looking forward to working with you. Best regards, Martin Yang
$140 USD em 7 dias
0,0
0,0

Ahmedabad, India
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