Voip line gone mute

[#1]

I can't see anything about Symmetric RTP

In that case , I'll leave Philip to sort you out with using STUN. Myself, I've never needed to use it...

[#2]

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@PhilipHeyes wrote:

Attached is a screen shot of our SPA112's RTP and NAT settings.

The RTP ports are important as these are accepting the in bound voice traffic during a call.

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My settings look the same as ours, except that the port min/max are 16384/16482

[#3]

I reset and made the necessary changes to make it connect again.  Still no luck.  I can hear the other person, but they can't hear me.

[#4]

Setup MicroSIP on a PC or laptop with the same type of SIP configuration and see if that gets 2 way voice.

[#5]

An incoming call seemed fine.  Outgoing still muted.  Might be random, of course.

[#6]

I'm reading this as MicroSIP has same one way voice issue as the Linksys ATA.

Did you mention there were Router Firewall settings that had been adjusted for UPNP & SIP ALG ?
We have SIP ALG disabled and UPNP Enabled.

By any chance do you have 2 Routers and a double NAT situation ?

Have you tried different RTP Ports ?

[#7]

Attached are my MicroSIP settings.  

The RTP port numbers must be set for the SIP session to connect & work i.e. they can not be left as all Zeros.

[#8]

Moderators Note

This topic has been moved from Home Phone to Tech Help

[#9]

I notice A&A appear to support G.711 A-law codec only.  

The SPA112 is a USA device so this is not the default out of the box, I would need to do the following :

Line1 / Line2 >>> Audio Configuration >
Preferred codec :  G711a
Secondary Preferred codec : Unspecified
Third Preferred codec : Unspecified


For MicroSIP,   Settings >>> Enabled Codecs would be just : G.771 A-law

[#10]

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@PhilipHeyes wrote:

I notice A&A appear to support G.711 A-law codec only.  

The SPA112 is a USA device so this is not the default out of the box, I would need to do the following :

Line1 / Line2 >>> Audio Configuration >
Preferred codec :  G711a
Secondary Preferred codec : Unspecified
Third Preferred codec : Unspecified


For MicroSIP,   Settings >>> Enabled Codecs would be just : G.771 A-law

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Yes, I have all that.  As I said, it worked fine (at least for out-going) for several months until this week.

[#11]

If the ATA and MicroSIP suffer the same one way voice issue it would point to an environment / network problem.

So what changed ?

e.g. the addition of another Router or a Wi-Fi Mesh or a VPN or enabling Parental Controls or enabling a DNS that has content filtering ?     Use of a PI-HOLE or similar advert blocker ?

Have Plusnet changed from using NAT on the Router to Carrier Grade NAT ?

Had anything else stopped working E.g Conf Calls / Messenger Calls / Wi-Fi Calling ?

[#12]

Have Plusnet changed from using NAT on the Router to Carrier Grade NAT ?

Plusnet never has, and AFAIK has no plans to use CGNAT

[#13]

At the moment I haven't changed anything in the router, except to experiment disabling uPNP.  I can't see anything about SIP ALG.

If do have one page for DMZ (currently disabled), which says:

DMZ
Only one device, with either a static or a DHCP address, can be placed into the DMZ. The Hub will give it a private IP address and forward all appropriate traffic to this device.

Important: Placing a host in DMZ has significant implications for its security. Although it will be still located behind the Hub’s firewall ALL unsolicited traffic not rejected by the firewall will be sent to this host by the Hub’s Network Address Translator, increasing it’s vulnerability to attack.

Would enabling this help?

[#14]

My VoIP connections work fine without using the DMZ.

This page has a link to a program to test for SIP ALG in your environment.


I notice in the A&A notes for the SPA112 the use of Default SIP Port 5060, for Sipgate this is not recommended.

So in our the ATA menu : Line 1 >> SIP Settings >> SIP Port : 47160

[#15]

oops .. here is the SIP ALG Detector ...

https://support.ringlogix.com/portal/en/kb/articles/sip-alg-detector