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one way audio with SIP trunk through a site-to-site VPN ...
https://community.cisco.com/t5/ip-telephony-and-phones/one-way-audio-with-sip-trunk-through-a-site-to-site-vpn-issue/td-p/4009491
one way audio with SIP trunk through a site-to-site VPN Issue. I have a problem that calls through VPN connection between two sites with sip trunk are established well (Signaling) but for media (RTP) they sometimes are one-way audio or no audio at all. There is a VPN connection between two sites by Fortigate.
One-way Audio over Site-to-Site VPN at Remote Sites ...
https://community.freepbx.org/t/one-way-audio-over-site-to-site-vpn-at-remote-sites/51675
However when I place a call from a Remote Site to the Main Site or another Remote Site, I see the Outbound SIP traffic (port 5060) go out over the Remote Sites WAN connection and not the VPN and there is no RTP port traffic displayed (hence the one-way audio).
SIP over VPN - One Way Audio (Incoming only) - Asterisk ...
https://community.asterisk.org/t/sip-over-vpn-one-way-audio-incoming-only/82022
I am now trying to place calls from the same PC (X-Lite) through my VPN and only have one way audio (incoming). I have added the following lines to sip.conf trying to get it working, but the problem remains the same: Blockquote [general] context=incoming externip=89.X.X.X ; VPN External IP Address localnet=10.X.X.X/255...
Solved: One-Way voice in a call over VPN - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/one-way-voice-in-a-call-over-vpn/td-p/2232262
One-Way voice in a call over VPN. We have a problem with the calls when they come from a user connected through VPN. The topology is the next: VPN User->ASA->ROUTER->User in office. Both users are in inside interface and the command " same-security-traffic permit intra-interface" is configured so the traffic is allowed.
One Way Audio Over VPN Issue - Endpoints - FreePBX ...
https://community.freepbx.org/t/one-way-audio-over-vpn-issue/73904
Hi All, I am having a “One Way Audio Issue Over VPN” and can’t seem to determine the fix. I’ve followed some of the other posts but they did not resolve my issue. FreePBX is running on 192.168.1.0/24. PJSIP extensions 101 and 105 on that network function great together. I then moved ext 105 over to 192.168.9.0/24, which is a wireguard VPN.
Experiencing One Way Audio When Connecting via SIP
https://knowledgebase.paloaltonetworks.com/KCSArticleDetail?id=kA10g000000PLooCAG
Experiencing one-way audio when connecting via SIP (Session Initiation Protocol). Environment PAN-OS Cause SIP (Session Initiation Protocol) allows two endpoints to establish media sessions with each other. This is an application layer signaling protocol. The main signaling functions of the protocol are as follows: – Location of an end point.
site to site vpn - One way audio on SIP phone on one call ...
https://serverfault.com/questions/15116/one-way-audio-on-sip-phone-on-one-call-direction-where-to-look-next
If it was a codec issue, you'd have no audio, because the phones would be unable to send audio between each other. One way audio is almost always caused by RTP not passing through. Per DLux's statement, turning off SIP ALG or SIP Fixup or SIP Transformation - different routers use different terms for the same thing - is a good first step.
How to resolve one-way or no-way audio on VoIP calls
http://info.teledynamics.com/blog/how-to-resolve-one-way-or-no-way-audio-on-voip-calls
One way to bypass the issues of NAT is to create a Virtual Private Network (VPN) on the enterprise network that places all the physical sites on the same internal subnet, or a series of linked internal subnets. VPN tunnels assist in passing packets from site to site, thus reducing the involvement of NAT on the edge router.
VPN One-Way Audio | 3CX Forums
https://www.3cx.com/community/threads/vpn-one-way-audio.73363/
42. Reaction score. 6. May 14, 2020. #1. Hello, I'm having trouble with some 3CX Desktop Client users who switch between HomeOffice and normal In-Office. When at home, they experience one-way audio while connected to the VPN and cannot hear the caller. Their audio is transmitted just fine, and the On-Premise 3CX receives external caller audio.
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