Having problems with inbound calls. When I receive a call, the caller can’t hear me, or I have severe broken audio.
I use Comcast and have Performance+ and Powerboost. I have an ASUS RT-N16 using DDWRT. I have tried with QOS and without QOS on for SIP and RTP… no luck.
I’ve turned on SIP Logging/Debug and the info is here:
Any help is appreciated.
SIP doesn’t carry the voice, it’s only signalling.
Nothing in a log is going to show quality. You need to measure jitter and packet loss between you and your carrier.
mtr is a good tool for this.
Ok, here is info from MTR… does this help?
|------------------------------------------------------------------------------------------|
| WinMTR statistics |
| Host - % | Sent | Recv | Best | Avrg | Wrst | Last |
|------------------------------------------------|------|------|------|------|------|------|
| 10.0.0.1 - 0 | 40 | 40 | 0 | 0 | 0 | 0 |
| 24.2.64.1 - 3 | 37 | 36 | 11 | 44 | 112 | 35 |
|te-4-1-ur07.saltlakecity.ut.utah.comcast.net - 0 | 41 | 41 | 9 | 24 | 87 | 9 |
|te-9-4-ar01.saltlakecity.ut.utah.comcast.net - 0 | 41 | 41 | 8 | 26 | 96 | 11 |
|te-9-3-ar02.saltlakecity.ut.utah.comcast.net - 0 | 41 | 41 | 9 | 26 | 87 | 10 |
|te-0-1-0-4-cr01.denver.co.ibone.comcast.net - 0 | 41 | 41 | 21 | 37 | 86 | 37 |
| xe-9-2-0.edge3.denver1.level3.net - 0 | 41 | 41 | 20 | 35 | 86 | 30 |
| vitelity-co.edge3.denver1.level3.net - 0 | 41 | 41 | 34 | 59 | 251 | 43 |
| No response from host - 100 | 8 | 0 | 0 | 0 | 0 | 0 |
| 66.241.99.21 - 0 | 41 | 41 | 34 | 49 | 89 | 35 |
| 64.2.142.18.gige-net.vitel.net - 0 | 41 | 41 | 35 | 48 | 88 | 35 |
|________________________________________________|______|______|______|______|______|______|
WinMTR v0.92 GPL V2 by Appnor MSP - Fully Managed Hosting & Cloud Provider
The latency is moving around a bit, but it is not that bad.
Keep in mind that you are testing to the Vitelity SIP server. The audio comes from whatever media gateway you are calling via.
The audio is consistently bad?
Yes, and consistantly when someone calls ME… They can’t hear me… but I can hear them just fine.
Are you sure you have your NAT setup right in Asterisk (SIP settings module in FreePBX)?
It was wrong… It was still on old network… (I had 192.168.0.x /24 and am now on a 10.0.0.x /23) But after correcting that no change.
It should be noted that I had this issue when I was on the 192 network as well.