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Audio issues - dropped packets


#1

This is a continuance of a previous thread that was closed since a new reply wasn’t added within 7 days. - Help with network setup - audio breaks up

To summarize, I’m dropping packets and getting an audio break-up. Here is a sample of my RTP Debug

Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018288, ts 64013128, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018289, ts 64013288, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018290, ts 64013448, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018291, ts 64013608, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018292, ts 64013768, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018293, ts 64013928, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018294, ts 64014088, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018295, ts 64014248, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018296, ts 64014408, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018297, ts 64014568, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018298, ts 64014728, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018299, ts 64014888, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018300, ts 64015048, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018301, ts 64015208, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018302, ts 64015368, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018303, ts 64015528, len 000160)
Got RTP packet from yyy.yyy.yyy.yy:13008 (type 00, seq 018304, ts 64015688, len 000160)

This seems to only be happening on the phones that are at our remote office locations. The PBX server is at our main location, and the phones work well there. Since the end of my last thread, I replaced the routers at one of the remote offices and the office with the PBX with Netgear Nighthawk X6 AC3200 units. I have QoS set up on these routers to grant all my phones and PBX server with the highest priorities. We are still experiencing problems.

1.Could this be caused by audio codecs? I just had ulaw selected in my asterisk SIP settings but my trunk has 'allow=ulaw&alaw". Is there a preferred codec to use?

  1. The Sangoma S500 phones at the remote office go into an unmanaged switch then into the Netgear router. The cable modem is bridged into the router. Could the switch or even modem be causing an issue? The setup is similar at the office with the PBX, but the signal goes bridged modem>Netgear router>PBX server. No switch is involved.

  2. Could the network card in my PBX server be an issue? I’m using the onboard ethernet port on the motherboard.

This is happening at multiple remote offices (I previously thought it was only happening at one office), so it leads me to think my issue is with the set up at the office that houses the PBX server.


(Tom Ray) #2

@Bradbpw So this debug doesn’t show Asterisk sending any RTP packets out. This shows Asterisk receiving them but is this from the phones IP or from the Providers IP?

What is exactly happening again? New thread, new details. Are you having issues hearing the inbound audio? Does the other said have issues with your outbound audio? And does this only happen on inbound calls or outbound calls (to the PSTN)?


#3

Sorry, the RTP debug I posted was a snippet from the full thing. It is sending packets as well.

xxx.xxx.xxx.x is the remote office IP where the phone is located. yyy.yyy.yyy.yy is the IP of my trunk provider

> Sent RTP packet to      xxx.xxx.xxx.x:12112 (type 00, seq 022774, ts 325280, len 000160)
> Got  RTP packet from    yyy.yyy.yyy.yy:17224 (type 00, seq 002037, ts 325920, len 000160)
> Sent RTP packet to      xxx.xxx.xxx.x:12112 (type 00, seq 022775, ts 325440, len 000160)
> [2018-10-19 15:28:09] ERROR[13806][C-00000055]: pbx_functions.c:701 ast_func_write: Function AUDIOHOOK_INHERIT not registered
> Got  RTP packet from    yyy.yyy.yyy.yy:13008 (type 00, seq 018288, ts 64013128, len 000160)
> Got  RTP packet from    yyy.yyy.yyy.yy:13008 (type 00, seq 018289, ts 64013288, len 000160)
> Got  RTP packet from    yyy.yyy.yyy.yy:13008 (type 00, seq 018290, ts 64013448, len 000160)
> Got  RTP packet from    yyy.yyy.yyy.yy:13008 (type 00, seq 018291, ts 64013608, len 000160)
> Got  RTP packet from    yyy.yyy.yyy.yy:13008 (type 00, seq 018292, ts 64013768, len 000160)
> Got  RTP packet from    yyy.yyy.yyy.yy:13008 (type 00, seq 018293, ts 64013928, len 000160)
> .....

The audio breaks up and the person on the other end says that they are having trouble hearing us. We sound slurred at times or words are missing altogether. It doesn’t always happen, but it will happen 1-2 times on maybe 30% of the calls. I know this happens on inbound calls, I think it’s also on outbound, I’ll try to confirm that.


(Tom Ray) #4

OK, let’s try to keep some clarification here. Because ext 100 (from problem location) to ext 200 (other location, could have issues too) is both an inbound/outbound call on that PBX. So right now let’s stick will calls to and from the PSTN.

So when a user at the remote location receives a call from the PSTN, the caller (PSTN side) is hearing the symptoms you described?


#5

Sorry for the slow reply. I have a new baby at home so I haven’t been in the office.

So when a user at the remote location receives a call from the PSTN, the caller (PSTN side) is hearing the symptoms you described?

Correct. The user at my remote location will also hear the symptoms coming from the caller on the PSTN side.

Is there any chance my network card could be causing the issue? I’m just using the built in motherboard Ethernet port.


#6

I just had a 4 minute internal (ext-to-ext) phone call between myself at the main office and a user at the remote office. The audio quality was great with no break-ups. So, does that point to my connection to the PSTN being the issue? If I had a bad connection to the PSTN would I see the dropped packets in my RTP debug like that? I
m using 1-voip as a trunk provider to the PSTN.


(Tom Ray) #7

Where they over the VPN?


#8

I don’t have a VPN set up between offices. I have tried in the past to get it set up and never had any luck with it.


(Tom Ray) #9

The audio issues could be a variety of thingsm poor bandwdith, poor connectivity, NAT, the router/network (either side) being overloaded…

What are all the common factors at each remote location that has this issue? Routers, switches, phones, etc…


#10

I should have plenty of bandwidth and the connection passed a VoIP quality test. Both locations have the ISP and service level.

Both locations also have the same router (Netgear AC3200). Previously, both locations had Asus NT66U routers, I replaced both with the Netgear routers while troubleshooting. The remote location phones run through an unmanaged switch. At the primary location where the PBX server is, everything is hooked directly into the router.

I’m starting to think it’s an issue with my PSTN connection since internal calls between offices are fine. I’ve created a support ticket with my trunk provider (1-Voip) to see if they can help me fix the issue.


(Tom Ray) #11

Well 41ms of Jitter is not great. You can start seeing problems at over 30ms. You will need to run one of these when you are actually experiencing the problem. Does any of those results change? Particularly, the Jitter or Latency.

This is the downside to intermittent issues, you have to do a bunch of things but you can only do them when it is happening.


#12

I ran a more comprehensive test using the Ring Central software. Everything looks to be good.

Untitled


(Tom Ray) #13

Again, you will need to run the test when the issue presents itself and see if it changes. If it’s not that then it’s going to point more to NAT.

Are you saying that internal calls, even from the remote locations, are no longer having issues?


#14

The issue happens pretty much all the time. I’d say there is an issue on 80% of the calls. Any call over 5 minutes is guaranteed to have an issue. The severity of the issue is the only variable.

Internal calls between remote locations (where the PSTN is not used) do not suffer from poor call quality. I have “Disable SIP ALG (NAT)” checked on both routers.


#15

This sounds most probably like a transient network degradation. Once the connection between FreePBX and VoIP provider degrades, you start having issues.


(Tom Ray) #16

Except the OP just clarified that this issue happens with remote phones at remote offices for internal calls. The VoIP provider isn’t involved in those. The only thing that is, their Internet connect and network at the main office where the PBX is.

The speed tests show jitter ranging from 30ish to 40ish ms and that’s at the high end of being OK. so again, there needs to be more testing run when the calls have completely degraded.


#17

Sorry for the confusion. In one of the OP’s post, it was said that the PSTN calls were affected but not the internal calls. In any case, the connection that links the local office with the remote office and the connection that links the local office with the provider are being affected. Maybe the connection is the same for both links.


(Tom Ray) #18

I get it. That’s why I had asked for clarification as well and wanted to confirm it wasn’t just the PSTN. Once the OP said still happens with “local” calls, that ruled out the provider.


#19

Except the OP just clarified that this issue happens with remote phones at remote offices for internal calls.

Sorry if there was any confusion, but that is incorrect. Internal calls between offices where the PSTN is not involved are perfectly clear and have no audio issues.


(Tom Ray) #20

OK then, that does change it. What codecs are the phones using and what codecs is the provider trunk allowing to be used?