Resolve 3CX WhatsApp Call Audio

@farooqtahir2013

As a seasoned Network, Cybersecurity, VoIP, and System Engineer with over 10 years in the field, I am confident in my ability to significantly improve your 3CX instance's WhatsApp call audio. Having worked extensively with Asterisk PBX, PJSIP, and Linux - all tools you mentioned that I will need to utilize for this task - I am well-versed in troubleshooting issues similar to the one you are experiencing. My proficiency with Wireshark, sngrep, and the 3CX console will allow me quickly trace the SRTP-RTP path and pinpoint the root cause of the one-way audio problem. Furthermore, my experience in working with a wide variety of vendors including Cisco and Fortinet enable me to cognitively handle multiple aspects of your project. To guarantee your satisfaction, apart from resolving the issue and providing you with in-depth documentation detailing all the changes made allowing seamless replication in the staging environment, I'm also available for a quick walkthrough or screen-share session to explain everything. Completing projects successfully is not just my objective; it’s an expectation I always aim to surpass. Let's get this issue resolved!

@AZTURK1

Hi there, I understand your cloud 3CX→Asterisk PJSIP setup where SRTP↔RTP conversion causes one-way audio; I can trace media, fix PJSIP/SRTP/NAT settings and deliver a reproducible patch with confidence. - Capture and correlate call flows (Asterisk pjsip, 3CX console) with sngrep/Wireshark - Identify SRTP→RTP translation issue, adjust PJSIP encryption/codec and NAT/STUN settings - Validate two-way audio across 5 consecutive outbound WhatsApp calls and provide configs pushed or delivered - Deliver a short written walkthrough and optional screen-share to replicate in staging Skills: ✅ Asterisk PBX ✅ PJSIP ✅ SRTP / RTP troubleshooting (Wireshark, sngrep) ✅ 3CX console integration and config deployment ✅ NAT/STUN and codec negotiation for reliable two-way audio Certificates: ✅ Microsoft® Certified: MCSA | MCSE | MCT ✅ cPanel® & WHM Certified CWSA-2 I can start immediately and confirm working calls within the first session; Do you already have pcap/sngrep captures for a failing call, or should I collect fresh traces once I get SSH and 3CX console access? Best regards,

@usamae

I can fix this quickly. The issue is almost always in SRTP to RTP negotiation or NAT handling on one leg, and your symptom matches that exactly. Plan: - trace both media legs with sngrep and RTP debug - verify PJSIP transport, rewrite_contact, rtp_symmetric, force_rport - align codec and crypto profile between Asterisk and 3CX - patch config, restart safely, and run 5 back to back call tests I have done similar 3CX + Asterisk voice routing fixes before. I will also send a short change log so your team can repeat it in staging. Can start right away, should be done fast if access is ready tbh.

@nolk

Hi there, I've been working in VoIP field for 10+ years and have lots of experience with commercial and freeware solutions. I'll be happy to help you with the setup and one-way audio issue.

@proggon

Hi there, I understand that you’re facing a challenge with ensuring two-way audio in your cloud-based 3CX instance integrated with Asterisk for WhatsApp voice calls. Specifically, it seems that the caller can’t hear the recipient, and resolving this issue is crucial for your communication workflow. To address this, I propose a detailed analysis of the SRTP to RTP path while also examining the PJSIP, codec, and NAT/STUN settings involved. By utilizing tools like Wireshark and sngrep, I will trace the audio streams and identify the root cause of the silence on one leg. After successfully establishing clean two-way audio through thorough testing on multiple outbound WhatsApp calls, I will compile a concise document that outlines the changes made to the PJSIP and SRTP configurations. This will help you replicate the fix in your staging environment seamlessly. Could you share more about any specific codecs you are currently using? https://www.freelancer.com/u/proggon Best regards, Wahaj Barlas.

@billybryan98

Hello I hope you are doing well. I am a VoIP engineer with deep hands-on experience in Asterisk, 3CX, PJSIP, and cloud deployments. I don’t rely on theory alone , I’ve routinely traced SRTP to RTP paths in cloud setups, adjusted media direction and NAT traversal settings, and tuned codecs to ensure reliable two-way audio for WhatsApp-integrated voice paths. In past engagements I’ve isolated silent-leg issues by inspecting signaling and media paths end-to-end, applying targeted PJSIP and SRTP tweaks, and validating changes with live captures from Wireshark, sngrep, and the 3CX console. I’ve delivered precise config changes and documented steps so teams can replicate them quickly, without breaking existing calls. I can handle this work end-to-end based on my experience: identify the silent leg, implement robust PJSIP/SRTP/NAT adjustments, verify two-way audio on several outbound WhatsApp calls, and provide a concise write-up you can drop into staging. I guarantee clean two-way audio and clear, reproducible documentation . Please feel free to contact me so we can discuss more details. I am looking forward to the chance of working together. Best regards, Billy Bryan

@naseeb1995

As an AI and Automation Specialist, my knowledge of intricate digital systems has given me a unique set of skills that include Software Architecture and VoIP. With over 10 years of experience, my professional journey has been laser-focused on designing intelligent systems that not only operate on autopilot but also reduce costs while improving efficiency - qualities that I'm proud to bring to your project. My familiarity with tools such as Asterisk 20+, PJSIP, Wireshark, sngrep, and 3CX console will prove instrumental in helping resolve the issue with your 3CX WhatsApp Call Audio. I understand the urgency of getting this resolved for you, and promise to meticulously trace the SRTP→RTP path and identify the root cause of the one-sided audio issue. Subsequently, I will precisely adjust the PJSIP, codec, or NAT/STUN settings to ensure clean two-way audio communication. Additionally, my commitment towards replicability is unwavering. Once I have successfully resolved this issue, I assure you a concise write-up of all modifications made, ensuring you're equipped to replicate these changes in a staging environment without any hassle. I offer not just hands-on expertise on the subject matter but also additional guidance for better comprehension, be it via quick walkthroughs or screen-shares – all aimed at providing sustainable solutions for you. This project presents an incredible opportunity for us to work together and ensure t!.......

@Hemantt841

I understand that you need to resolve the audio issue with your cloud-based 3CX instance, specifically ensuring that both the caller and recipient can hear each other during WhatsApp calls. The task involves tracing the SRTP to RTP path and adjusting the relevant PJSIP, codec, or NAT/STUN

@tanaa4

Hey there, I can fix this. I have deep hands on experience with 3CX, Asterisk 20 plus, PJSIP, and real world one way audio issues in cloud only SIP setups. Your symptoms point to an RTP and SRTP negotiation or media routing mismatch, usually caused by incorrect SDP address selection, RTP port reachability, codec mismatch, or SRTP policy differences between legs. My approach: 1. Capture a failing call with sngrep and a packet trace to verify SDP offers and answers, chosen codecs, and where RTP is actually sent 2. Confirm whether 3CX is sending RTP to the correct public IP and ports, and whether Asterisk is decrypting SRTP correctly and re injecting RTP back to 3CX 3. Fix media settings: external media address, local networks, rtp symmetric, rewrite contact, force rport, ICE support, STUN where needed, and align SRTP policies 4. Lock down codec set and DTMF mode to remove negotiation edge cases 5. Validate with five consecutive outbound WhatsApp calls and provide evidence Deliverables: Updated PJSIP and media configs, a short replication write up for staging, and a quick screenshare walkthrough. If you share the current pjsip conf and 3CX trunk settings, I can start immediately.

@SnTechnologies59

Hello Sir, I have 7+ years of experience with 3CX, Asterisk (PJSIP), SRTP/RTP, and Twilio/Telnyx voice integrations. I understand one-way audio issues in cloud VoIP environments and can trace the SRTP→RTP media path, identify the root cause, and restore clean two-way audio. I can resolve this immediately and am waiting for your response. Best regards, SoftNexus Technologies

@slavkom1

Hi there. I get what you’re facing with the 3CX to Asterisk setup and the one‑way audio on WhatsApp calls. This is usually a media flow or crypto mismatch, and I’ve dealt with similar RTP/SRTP issues before in cloud-only VoIP stacks. I can dig into it by checking: • PJSIP media settings and SRTP profiles • How 3CX is advertising RTP candidates • NAT, STUN, and ICE handling on both sides • Live captures in sngrep or Wireshark Once we spot the broken leg, I’ll correct the configs and keep notes so you can mirror them in staging. I can start today and this shouldn’t take long once I have SSH. Before I jump in, can you confirm whether Asterisk is terminating or relaying SRTP on this trunk? Greetings, Slavko

@ayesha86664

Hi there! It’s frustrating when WhatsApp calls connect but only one side is audible—this breaks workflow and affects communication reliability. Getting clean two-way audio on your 3CX-Asterisk setup is essential. I have hands-on experience troubleshooting VoIP audio issues over PJSIP and SRTP, configuring 3CX and Asterisk for cloud deployments, and resolving NAT/STUN and codec mismatches that commonly cause one-way audio. I’ve successfully traced RTP flows with Wireshark and sngrep to identify and fix silent-call problems in previous projects. My approach will be to methodically trace the SRTP→RTP path, adjust the relevant PJSIP and codec settings, test multiple calls to confirm two-way audio, and provide a concise walkthrough so you can replicate the fix in staging. Check our work https://www.freelancer.com/u/ayesha86664 Do you want me to also document the exact configuration changes for future deployments? Let me know if you’re interested & we can discuss it. Best Regards, Ayesha

@joeyboym

Your issue clearly points to an RTP/SRTP negotiation or NAT traversal problem between 3CX and Asterisk — most likely in the media path rather than signaling, since calls connect but audio is one-way. I’ve worked extensively with Asterisk (18–20+), PJSIP, SRTP bridging, and 3CX integrations in fully cloud-hosted environments. One-way audio in SRTP→RTP scenarios typically stems from incorrect media_address settings, asymmetric NAT handling, codec mismatches, or SRTP policy conflicts between legs. My approach will be structured and fast: • Capture and inspect SIP/RTP flows using sngrep and Wireshark • Validate PJSIP endpoint, transport, and SRTP profiles • Verify NAT, rewrite_contact, rtp_symmetric, force_rport, and external IP configs • Confirm codec alignment and RTP port ranges • Trace the exact SRTP→RTP bridge inside Asterisk Once isolated, I’ll adjust the necessary PJSIP, NAT/STUN, or codec parameters and test until we confirm stable two-way audio across at least five consecutive outbound WhatsApp calls. Deliverables: • Clean, working PJSIP/SRTP configuration • Summary write-up explaining root cause and applied fix • Short walkthrough / screen-share session SSH access is perfect — I can begin tracing immediately. Let’s get your media path fully symmetrical and production-safe.

@Ammad026

I can troubleshoot and fix the one-way audio issue on your cloud-based 3CX outbound WhatsApp calls. I’ll trace the SRTP→RTP path, analyze codecs, NAT/STUN settings, and PJSIP configurations to pinpoint the problem. Once identified, I’ll adjust the relevant settings to achieve clean, reliable two-way audio. I’ll validate the fix across multiple test calls to ensure consistent results. Afterward, I’ll provide updated configurations and a short walkthrough so you can replicate the setup in staging. Ready to start immediately and ensure your WhatsApp voice integration works flawlessly.

@dejanv7

Hi there, I’ve taken a close look at your scenario: a cloud-based 3CX setup talking to WhatsApp via the PJSIP/WhatsApp API where the caller can hear every time but the recipient cannot hear the caller. You’re on Asterisk 20+ with SRTP from Asterisk and RTP from 3CX, and the task is to trace the SRTP→RTP path, identify the silent leg, and adjust PJSIP, codec, or NAT/STUN settings to achieve reliable two-way audio. I’ve worked on several VoIP stacks with similar two-way audio challenges in cloud deployments, including tracing SRTP/RTP paths and tuning PJSIP profiles for NAT traversal in cloud-only environments. I would approach this by instrumenting the media path end-to-end: verify encryption translation points, capture RTCP to assess packet loss and jitter, and compare negotiated codecs across both legs. I’d adjust PJSIP transport and SRTP profiles to ensure sequence numbers and timestamps align, review NAT bindings and STUN usage, and test with the WhatsApp outbound flow to confirm bidirectional audio. I’ll document the exact changes in a compact write-up suitable for staging so you can replicate them. Best regards, Dejan