[SOLVED] 1-way audio issues with Asterisk 16

FreePBX Providers
1

Hi, first would like to say that I did a search in this forum for audio issues with Asterisk 16 and found none. since yesterday, have been experiencing 1-way audio issues. First noticed it when calling a place which I could hear but they could not hear me. Called the same DID 10x within a few minutes. Same 1-way audio each time. They decided to call me back (used my personal DID to call so when they call back there is no IVR) and again could hear them but they could not hear me.

A few minutes ago, someone on the same FreePBX server as I am on called me. No audio period. They called back again a few minutes later. This time was logged into CLI and watched RTP and see it as one-way and in the middle there is a NAT DEBUG notice about changing IP addresses. The RFC1918 address is behind the router at xxx.252.159.31 :

Sent RTP packet to xxx.252.159.31:60996 (type 00, seq 019059, ts 041142, len 000170)
Sent RTP packet to xxx.252.159.31:60996 (type 00, seq 019060, ts 041302, len 000170)
Sent RTP packet to xxx.252.159.31:60996 (type 00, seq 019061, ts 041462, len 000170)
[2019-12-27 14:11:51] DEBUG[16828][C-00000060]: res_rtp_asterisk.c:7358 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 192.168.25.72:60996
Got RTP packet from 192.168.25.72:60996 (type 126, seq 012373, ts 000641, len 000010)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019062, ts 041622, len 000170)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019063, ts 041782, len 000170)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019064, ts 041942, len 000170)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019065, ts 042102, len 000170)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019066, ts 042262, len 000170)
Got RTP packet from 192.168.25.72:60996 (type 00, seq 012374, ts 041440, len 000170)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019067, ts 042422, len 000170)
Got RTP packet from 192.168.25.72:60996 (type 00, seq 012375, ts 041600, len 000170)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019068, ts 042582, len 000170)
Got RTP packet from 192.168.25.72:60996 (type 00, seq 012376, ts 041760, len 000170)
Sent RTP packet to 192.168.25.72:60996 (type 00, seq 019069, ts 042742, len 000170)

During this call, I know for sure voicemail greeting came on because I know how many rings it takes before voicemail is activated. There was no audio heard meaning I could not hear the persons voicemail greeting.

After the above call ran # asterisk-version-switch and downgraded asterisk to Asterisk 13.29.2

Call the persons extension after downgrade of Asterisk was finished. The person did not answer but I did hear voicemail greeting. The rtp debug shows what I consider normal packet flow for audio and CLI only shows the RTP packets once vmail greeting started:

Got RTP packet from 192.168.25.21:12406 (type 00, seq 007960, ts 021600, len 000160)
Sent RTP packet to 192.168.25.21:12406 (type 00, seq 008051, ts 021600, len 000170)
Got RTP packet from 192.168.25.21:12406 (type 00, seq 007961, ts 021760, len 000160)
Sent RTP packet to 192.168.25.21:12406 (type 00, seq 008052, ts 021760, len 000170)
Got RTP packet from 192.168.25.21:12406 (type 00, seq 007962, ts 021920, len 000160)
Sent RTP packet to 192.168.25.21:12406 (type 00, seq 008053, ts 021920, len 000170)
I am at a loss as to what is causing this audio issue. Find it hard to believe Asterisk 16 is the cause. But fact remains with Asterisk 13 this audio issue is not reoccurring so far.

Thanks.

2

Even though one version works and the other doesn’t, this seems to be a NAT issue. Is your FreePBX behind a NAT device? If so, have you configured NAT parameters on SIP settings and forwarded the corresponding ports on the NAT device?

3

Hi, PBX is in data center with static IP. The endpoints are all remote going through router-to-router vpn. I did see in the rtp packets the public address of the router which is remote to the PBX. The packets should be going only to RFC1918 address.

Below are 2 endpoints below are behind the 99. address
Contact: 308/sip:[email protected]:5160 266cb6fd96 Avail 39.423
Contact: 308/sip:[email protected]:5165 4503c2e770 Avail 48.579

4

Make sure that you correctly set the NAT parameters in SIP settings and that you also correctly define the local networks.
The network in the remote site must be defined as a local network on FreePBX too.

Is 308 an extension? If so, why does it appear connected from two different IPs? Are you using the contact PJSIP functionality?

Depending on the endpoint, you might need to set the extension to nat=yes even though you are connecting through a VPN. For example csipsimple softphone needs it.

5

As mentioned at the beginning of this thread, could not find any other post with 1-way audio issue with Asterisk 16. Finally found the cause. During the last couple of days while away from the office, the Bria Mobile which is used a lot constantly had 1-way audio problem. What was noticed was when using external speaker instead of the ear speaker, the 1-way audio was non existent. So it turns out there is something wrong with Phone itself. It’s acting like the MIC is muted permanently when using the phone’s internal speaker.

6

That setting is irrelevant. It’s a Chan_SIP setting not a PJSIP setting.

7

Sorry, not as much experience with PJSIP as with chan_sip.

8

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.