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The causes of No-Audio and One-Way-Audio VoIP Calls ...
https://blog.kolmisoft.com/the-causes-of-no-audio-and-one-way-audio-voip-calls/#:~:text=A%20firewall%20can%20be%20misconfigured%20to%20block%20audio,of%20one-way%20and%20no-way%20audio%20on%20VoIP%20calls.
SIP and NAT: Why is it a problem? – The ... - Smartvox
https://kb.smartvox.co.uk/voip-sip/sip-nat-problem/
The audio streams use the RTP protocol and they are generally established directly between end-points using different ports to those used for the SIP messages. The RTP ports go into a “listening” state whereby they can accept a new connection from a remote device.
Audio - RTP Issues - Installation / Upgrade - FreePBX ...
https://community.freepbx.org/t/audio-rtp-issues/20771
Send output of NAT section of sip show settings command. Send output of sip show peer xxxx where xxxx is the iPhone peer, again only need the NAT secton. Lastly make a phone call with no audio and issue the rtp set debug on command and see what IP the server is sending the audio to and see if you are getting any audio.
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 of the most common challenges involves a technology called Network Address Translation (NAT), which for data networks has been a godsend, but if not configured carefully, could cause problems for voice applications. In particular, NAT is a common cause of one-way and no-way audio on VoIP calls.
nat - Audio issues with asterisk 13 - Stack Overflow
https://stackoverflow.com/questions/39684843/audio-issues-with-asterisk-13
Show activity on this post. thanks, idescovered the problem and its solution. first of all, the wright NAT options are: nat=force_rport,comedia. second, the wright media option is: directmedia=no. my problem was related to opened port, in …
The causes of No-Audio and One-Way-Audio VoIP Calls ...
https://blog.kolmisoft.com/the-causes-of-no-audio-and-one-way-audio-voip-calls/
For a voice call to function properly, the RTP packets must travel from one side to the other, through the firewall/NAT. A firewall can be misconfigured to block audio traffic. Devices under NAT could be unable to bypass NAT properly. A device could be unaware of the NAT’s existence and send its local IP to the Internet.
Audio is being stripped from RTP stream from VoIP Provider ...
https://forum.netgate.com/topic/95976/audio-is-being-stripped-from-rtp-stream-from-voip-provider
I still receive the RTP stream when I have no sound on the call. In the last instance, I recieved 1293 RTP packets from the SIP provider, however the payload was all d5's and there was no sound at all. Surely if this was a NAT issue the RTP stream would not be reaching the PBX at all? RTP is media and SIP is signalling.
A tale of VoIP, NAT and some confused engineers | …
https://blog.apnic.net/2020/09/29/a-tale-of-voip-nat-and-some-confused-engineers/
When a VoIP device is behind a NAT, the IP and port that it puts in SDP are usually wrong as the NAT router will change these when the RTP packets leave the network. As such, a common mechanism used by VoIP Service Providers is to wait for some RTP packets to be received from the remote VoIP device and use their source IP and port in preference ...
How to Troubleshoot VoIP Issues with Palo Alto Networks ...
https://knowledgebase.paloaltonetworks.com/KCSArticleDetail?id=kA10g000000CmUiCAK
ALG is supposed to translate them to the public IP as per the NAT rules configured. This is important. Otherwise, the RTP communication will not work resulting in audio or video issues. For SIP, check the SDP Payload in SIP Invite and SIP 200 OK packets. They contain the IP address for RTP in Connection Header and Ports in Media:
More RTP Issues | Netgate Forum
https://forum.netgate.com/topic/27359/more-rtp-issues
I have a problem with RTP traffic not flowing through through the NAT from WAN->LAN. RTP traffic going from the LAN->WAN works fine, and we have audio from the LAN VoIP server out to the WAN VoIP server. The problem is that RTP packets are arriving at the pfSense firewall WAN port but are not being forwarded to the LAN, and I can't figure out why!
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