No audio on softphones, but rings


(Theta Coder) #1

First off, thanks for being here, this is a great place for information.

This issue is a result of the wonderful health hysteria currently happening. We are having to consider having our techs work remotely from home. We presently run Yealinks internally on CHAN_SIP and want to put in a set of softphones on home PCs/Mobiles. We have created secondary extensions for these softphones.

Our Phones run on a dedicated 10.x network to the EGS Gateway unit owned by the SIP Trunk provider. The Main FreePBX has dual NIC, one fo rthis phone network, and one for the business network so we can admin and use voice mail etc. Now I am testing on my Android using Mizudroid. We setup a firewall rule (SonicWALL) for 5160/5161 and 10000-20000 UDP and a NAT policy to match to send those ports to the phone server.

I can get the softphone to register and authenticate via the internet and business IP (not the dedicated 10.x network), and we can even dial extensions between the desk phones and the softphone in both directions, but once connected, we get no audio. I checked the codec on the softphone and is matches the preferred one in the server.

We are running FreePBX 14.0.13.26 on Asterisk 16.6.2


#2

Confirm that your SonicWALL is set up correctly, especially Consistent NAT:

Confirm that in Asterisk SIP Settings, External Address and Local Networks are correctly set. If you change these, you must restart (not just reload) Asterisk.

Confirm that the extension has NAT Mode set to Yes, and that any NAT related settings in the client are turned off.

If you still have trouble, at the Asterisk command prompt, type
sip set debug peer xxx
where xxx is the extension number. Make a failing test call, paste the relevant section of the Asterisk log at https://pastebin.freepbx.org and post the link here. Also, report whether the desk phone user can hear the softphone user, and vice-versa.


(Theta Coder) #3

There were no NAT settings in the FreePBX.

I Detected the settings (it picked up the Business WAN IP addy fine, and added both internal networks), and will reboot the server after hours.

For the softphone extension I enabled NAT, and on the client disabled ICE and STUN, and there did not appear to be any NAT settings.

Consistent NAT is now enabled on the Firewall unit. I will do some more testing and is it still fails out, I’ll upload the logs and see what can be done.


(Theta Coder) #4

Placed a failed call this morning. Below is the pastebin for the output of the full log from when the softphone (ext 205) came online. I then tried calling it from my deskphone (ext 105)

https://pastebin.freepbx.org/view/a3aac248


#5

Unfortunately, the log does not contain a SIP trace. Did you get an error when you issued
sip set debug peer 205
?
Possibly, if you typed that before it came online, it did not recognize the address.
Give the command again and make another test call. If the trace does not appear in the log, try
sip set debug on
which will trace all SIP traffic on the system, then try again. After you get the log, do
sip set debug off
because on a busy system, all the SIP traffic can be a lot of data.


(Theta Coder) #6

My apologies, bit hectic here.

Here is the updated link with the SIP debug turned on. Called from the deskphone (ext 105) to the softphone (ext 205) Got ring, got successful connect, but no audio.

https://pastebin.freepbx.org/view/d384f062

And as near as we can tell neither end can hear the other speak.


#7

The log starts too late (with the ACK to the INVITE we are interested in). In Asterisk Logfiles, set Lines to 1000 and try again.


(Theta Coder) #8

Ok, set the log to 1000, waited till I had no other calls, and hopefully this one has all teh needed info

https://pastebin.freepbx.org/view/a8f68de5


#9

There was enough other noise that the INVITE to 205 and its responses are absent. In your first log, you see a line containing Called SIP/205 and one containing Spawn extension (ext-local, 205, 2) exited non-zero – these lines should be found in your log with the trace; paste everything between them.

However, there is a clue in what you posted. Line 109 has
Via: SIP/2.0/UDP 172.58.229.235:10182;branch=z9hG4bK-417p6632505073162535216r;received=172.58.229.235;rport=20946
which shows that the phone was NATted as expected by T-Mobile (the source port numbers are different), but that MizuDroid was attempting NAT traversal (public IP in the Via header).

I would expect that not to fool chan_sip into thinking there was no NAT, but there might be a bug.
Confirm that ext. 205 has NAT Mode set to Yes (force_rport, comedia).


(Theta Coder) #10

I have been looking through newer traces, and trying to get more than 30 seconds when no one is on the phones to get another trace, no luck so far.

I was unable to find those 2 find points in any of the logs, even with lines set to 1000. I went and checked all of the settings in MizuDroid and couldn’t find anything that looked out of place.

I know my firewall has 10000-20000 set as the valid range, so the port being 20946 would be outside that allowable range would it not?

I also tried Zoiper, and it too had the same behavior, will register and authenticate, but once connected, no audio. I will continue to try and get a good log.


(Theta Coder) #11

Finally got something. Here is the log, it contains from the Called SIP/205 to the end of the call…although the second part never seemed to show up

https://pastebin.freepbx.org/view/abcaf0df


#12

But the SIP trace is gone. The
sip set debug on
gets cancelled by a restart or reload. If you made any config changes since you gave that command, that would explain it. In any case, set the debug on right before you make the test call, confirm that the SIP trace is in the log, then paste it and post the link.


(Mulderlr) #13

If you are using T-mobile, it isn’t going to work using cellular data. They block RTP, so you will need to use a VPN and softphone together, or use wifi from a network that supports it. For example, I have a tmobile phone and use zoiper to my PBX fine over wifi, but over cellular, the calls ring, but no audio. Connect the VPN to the PBX, and its all fixed. I don’t need the VPN anywhere else or on wifi, so it is definitely an issue with Tmobile. Anyone NOT having this issue with tmobile LTE I would be curious to know about.


#14

No problem here, GSWave client using TCP:5060 using Volte on mintmobile (t-mobile msno).


(Mulderlr) #15

Hmm, perhaps it is market based or perhaps something has changed recently. I use TLS with SRTP and it has never worked. Today after I posted to this thread, it is magically working over cellular. If it breaks again, I might have to blame zoiper, but there are plenty of threads on reddit blaming tmobile, and since I had the exact same issues, I didn’t waste much time on it since a vpn was a simple enough work around.