I will troubleshoot sip rtp nat and voip call issues
FullStack SaaS, VoIP Engineer
Level 1
Has met certain performance criteria and shows strong potential in the marketplace.
About this Gig
Calls failing, dropping, or connecting without audio? I will analyze one defined SIP or RTP incident and turn the trace into a practical next step. I inspect SIP messages, response codes, dialog state, authentication, routing, SDP, NAT, codec negotiation, RTP paths, and relevant firewall behavior. My reference project includes a custom Go SIP routing server, SIP trace storage, sngrep-style call inspection, RTPengine control, dispatcher routing, registrar flows, and end-to-end SIP smoke tests. Basic delivers a written diagnosis; Standard includes one bounded fix; Premium follows one call path end to end. Results depend on the evidence and access available, and third-party carrier faults may require provider action. Message me before ordering if more than one environment, trunk, or symptom is involved.
Provider:
Zoom
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OpenPhone
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Webex Calling
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Cisco
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3CX
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Asterisk
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FreePBX
My Portfolio
FAQ
Can you guarantee a fix?
No. I can guarantee a structured investigation and honest findings; the fix may depend on access, a carrier, firewall owner, or unavailable evidence.
What counts as one incident?
One symptom in one environment on one call path, represented by a focused trace or log set.
Can you analyze one-way audio?
Yes, when the SIP/SDP and network or RTP evidence needed to isolate the media path is available.
Do you need server access?
Not for a trace-only diagnosis. A bounded fix may require temporary access through Fiverr's approved order workflow.
Can you work with a carrier issue?
I can prepare evidence and provider-facing questions, but I cannot control or guarantee a third party's resolution.
Will you inspect production safely?
Yes. I begin read-only and agree on any change and rollback before implementation.

