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FreePBX Voice and Audio cutting in and out
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FreePBX Voice and Audio cutting in and out

HuntersPadHuntersPad Member
edited September 2017 in Help

I moved to RamNode Atlanta VPS for my FreePBX server. My Previous VPS was in Washington,DC.

Since day one, Audio is randomly cut in and out even with VOIP to VOIP calls through my PBX. (Can provide a number via PM to show proof) Its not all the time but its scratchy to put it as in example. Words can be made out but it cuts in and out within a millisecond. This was not happening before on my older VPS. The VPS is more than capable hardware wise. I'm not blaming RamNode, just maybe I have a config error somewhere?

No packet loss on my home ISP and my average ping to my VPS is only 15ms.

Comments

  • asterisk14asterisk14 Member
    edited September 2017

    are all your ports forwarded/open that are in use such as RTP, UDP. RTP ones carry the audio so start there and make sure same ones are in use on your voip server and your voip adapter and are open/forwarded in your router to your voip adaptor.

    Check your router's SIP ALG settings.

    Was the freepbx installed by you or was the image provided by you/ramnode?

    Have you tried just installing asterisk on it's own to see if its an asterisk problem/vps problem or voip adaptor/router problem.

  • @asterisk14 said:
    are all your ports forwarded/open that are in use such as RTP, UDP. RTP ones carry the audio so start there and make sure same ones are in use on your voip server and your voip adapter and are open/forwarded in your router to your voip adaptor.

    Check your router's SIP ALG settings.

    Was the freepbx installed by you or was the image provided by you/ramnode?

    Have you tried just installing asterisk on it's own to see if its an asterisk problem/vps problem or voip adaptor/router problem.

    Provided by RamNode and it's the latest.

    I don't think port forwarding on my router will change much. As in my experience in the past if it was an issue with that it wouldn't work at all

    I'm using Bria on mobile and a Grandstream phone. Along with calls from a normal landline or mobile line also hear the studder so it's definitely the VPS side of things.

    The studder is not all too noticeable with normal voice calls but catch it here and there. But with Music on hold as an example is where you can really tell something is not right.

  • Not very familiar with PBX/SIP but have you tried playing around with the audio codecs maybe?

  • @Ishaq said:
    Not very familiar with PBX/SIP but have you tried playing around with the audio codecs maybe?

    Its not that high bw/low bw same issue.

  • If you're strictly in a scenario where it's from one phone to another then the media layer is acting as a packet forwarder only. This isolates things a bit to either the traffic itself having a lot of jitter and potentially being out of order, or the reading/sending of the traffic encountering a delay which introduces the jitter. If you do a packet capture of the traffic you should see an RTP packet being sent and received approximately every 20ms. Wireshark can also interpet and show you the jitter. I'd suggest giving that a go as it will help narrow things down.

    Thanked by 1HuntersPad
  • HuntersPadHuntersPad Member
    edited September 2017

    file said: rder only. This isolates things a bit to either the traff

    @file said:
    If you're strictly in a scenario where it's from one phone to another then the media layer is acting as a packet forwarder only. This isolates things a bit to either the traffic itself having a lot of jitter and potentially being out of order, or the reading/sending of the traffic encountering a delay which introduces the jitter. If you do a packet capture of the traffic you should see an RTP packet being sent and received approximately every 20ms. Wireshark can also interpet and show you the jitter. I'd suggest giving that a go as it will help narrow things down.

    Thanks!

    Is this helpful? https://s26.postimg.org/yi4c48li1/IMG_1550.jpg

  • If you look at the packet capture do you see anything of note when the audio cutout occurs?

  • @file said:
    If you look at the packet capture do you see anything of note when the audio cutout occurs?

    Nope, When it cuts out its only a ms like a quick skip. I may go back to my old provider and buy a new box and compare.

  • It may be small packet loss from your new server to you. Your link does show a packet lost.

  • If I decide today to move back to my old provider I'll provide an update. Only difference is my old provider I was running V13 of FreePBX and the current one now V14

  • @asterisk14

    Here is something interesting I wasnt aware of. Why is this running at port 5000? Is this normal? I just did a new install and found it. And sure enough its been running on my previous install as well. Is this supposed to be Bundled with FreePBX??!
    https://github.com/sdelements/lets-chat

  • @HuntersPad said:
    @asterisk14

    Here is something interesting I wasnt aware of. Why is this running at port 5000? Is this normal? I just did a new install and found it. And sure enough its been running on my previous install as well. Is this supposed to be Bundled with FreePBX??!
    https://github.com/sdelements/lets-chat

    No idea if that should be running or not. Last time I used FreePBX it was a while ago, now I just use asterisk on it's own as it requires less resources and therefore can be run on a 128MB VPS. Second reason is that as things get more complicated with GUI front ends, it introduces vulnerabilities and potential for glitches. Last time I used the Digium frontend, someone was able to repeatedly hack it and place many calls to the Zionist entity. Since then I don't use any front end now and asterisk is running pretty great with >120 days of uptime and more importantly no more hacks.

    That's why I said try running asterisk on it's own and see if the effect is the same.

    Can you record the audio and post it here for me to listen to? TBH this may be normal to some extent. I test my asterisk by generating an echo and a continuous tone from the server and I do notice that the continuous tone sometimes drops out for a fraction of a second. However this does not really affect my experience.

    I see you have a thread open on dslreports, they should be able to help you better than here as there are a lot of specialists on there with VoIP experience.

    Thanked by 1HuntersPad
  • Does plain Asterisk have the follow me feature like freepbx does?

  • Why don't you just install FreePBX 14 on top of a clean Debian image? It takes literally like 30 minutes to get installed and running, then you don't wind up with any bloatware or anything that the FreePBX image includes by default.

  • @HuntersPad said:
    Does plain Asterisk have the follow me feature like freepbx does?

    There is a Followme application in Asterisk which implements some functionality, but not knowing how FreePBX has done things they may have implemented some stuff in dialplan to do it instead.

    Thanked by 1HuntersPad
  • @file said:

    @HuntersPad said:
    Does plain Asterisk have the follow me feature like freepbx does?

    There is a Followme application in Asterisk which implements some functionality, but not knowing how FreePBX has done things they may have implemented some stuff in dialplan to do it instead.

    Thanks! any VPS providers have a image/iso for installing it? Or would I have to install it from scratch?

  • If you are referring to Asterisk itself we don't have an ISO or distribution for it. It's a piece of software you install, much like Apache or other things. It's only the FreePBX+Asterisk combos that have distros generally. Asterisk itself also does not include a GUI (in case you were unaware) - it's configured via text files by default.

    Thanked by 1HuntersPad
  • @file said:
    If you are referring to Asterisk itself we don't have an ISO or distribution for it. It's a piece of software you install, much like Apache or other things. It's only the FreePBX+Asterisk combos that have distros generally. Asterisk itself also does not include a GUI (in case you were unaware) - it's configured via text files by default.

    Thanks! Yeah I sort of thought it maybe config based by default. My NAS has an install for plain Asterisk I may install and play with (it has a GUI) But I'd never trust that in production lol.

    Update to my issue with the audio cutting in and out, I installed FreePBX 14/Asterisk 13 in a VM on my home server. When using it through LTE or call from a DID that call audio issue is not present. I may host it at home, I'm just reluctant to. (I do get two IP's from my ISP) so I could VLAN it for security.

  • The package you are referring to for your NAS is quite an old version, and the GUI has not been maintained for quite a few years. It likely suffers from security issues. I wouldn't recommend using it even for experimentation.

    Thanked by 1HuntersPad
  • Bookmarking this thread to see how it turns out; I've been PBX curious for several years. I don't want to setup anything big, but I wouldn't mind migrating from my existing VoIP provider. They work, but I'd prefer to control it myself.

  • @WSS said:
    Bookmarking this thread to see how it turns out; I've been PBX curious for several years. I don't want to setup anything big, but I wouldn't mind migrating from my existing VoIP provider. They work, but I'd prefer to control it myself.

    Yeah I like control! By the end of the year I'll most likely port my Home phone NetTalk line over to it pay by Min actually comes out cheaper than the $40 ish a year I pay for it.

    I get telemarketing calls from time to time that think they reached someones voicemail which causes them to hang up LOL. I also use it to play with people who call my number thats in a 1980's song.

  • You used your imagination, but was disturbed?

  • @WSS said:
    You used your imagination, but was disturbed?

    Disturbed is one way to describe some of the things callers hear...

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