Horrible quality with one phone

I have a an S700 sangoma phone. It’s never sounded good… I did a test call today and you can hear as if packets are dropping. We have a 50/5 internet and the system is in the cloud. The weird thing is, it did the same thing even when it was on premise. Codec issue? SIP ALG is off on the carrier and router.

What firmware are you using on the phone and what codex are you using.

There’re so many factors that could be causing your problem.

Please provide a scheme of your setup when I you had the PBX on premise.

Well I can’t imagine it’s the firmware because it does the same thing in all models of yealink as well. How do I check the codec? What is recommended?

Maybe you need to look into doing a little QoS on your internal network for your voice if all phone models are doing that? Then again if on premise calls are doing that, it could be something else, How did you do a test call, an echo test on the system, or a call via your carrier?

If it’s doing it in and out of the LAN, start with the obvious.

Replace the cable. Look at your port duplex settings, auto detect is great when it works. Check the switch port for errors.

It’s rare, but possible the nic in the phone is bad. (This coming from someone who has seen duplicate MACs on a /24 network twice in 15 years). The highly improbable does happen.

But the person says it happens on any phone including yealink at this location. This is 100% a network problem at that location.

seems that the issue is with sipstation and codec related. We have done everything we can think of. Turned off SIP ALG on the isp modem and router. Setup VLAN just for voice and QOS. Replaced the switch and router and still have the exact same issues at 3 locations. Not impressed with sipstation support either.

I did a trial of sipstation and wasn’t impressed with the latency either.

Primary sever was about 65ms and the secondary was around 35ms. You would think their software would default to the lower latency server.

Ended up just staying with my previous provider as it’s 5ms to them here in Toronto. Sangoma needs to peer with better providers.

we like freepbx for our solution because it’s simple. Our next phase is to change sip providers in our home office. We are in ohio. Suggestions?

VoIP.ms and choose premium routing.

I used them in the past on an on premise system. I’ll try some testing

With any provider, including our SIPStation product the latency to the server you register with does not effect call quality.

Those servers are for registration and call origination. The media is handled directly to the gateway closest to the LATA you are calling.

If you are restricting UDP traffic to just the SIPStation registration servers that could be the issue.

Is the PBX on premise? You make mention of hosted but I was not clear. If hosted I suggest you try the VPN built into the phone to take your LAN and firewall out of the equation.

How about you get the facts before spreading FUD. The trunk servers are just for signalling. Not media. All media is direct peered with top level providers only and media goes to closest pop to where you are calling to make latency issues a non factor and would be useless to pick the closest server. This is how premium provides work. Providers that handle media on the same server are just asking for problems and introduce way extra call delay throughout the call. You may not notice it as bouncing latency would be but think of all the extra hops that media/RTP has to take to complete a voice packet.

Secondly asterisk has no ability to weight trunk order by latency. If it did we would use it. You control which trunk you use in your outbound routes. By default they try trunk1 first then trunk2.

Sorry to sound like a jerk but when advising people in things make sure you have facts first and don’t state things as facts that clearly you are not in the know on.

What codec are you using. I would stick with ulaw. If trying to do g729 then you will need a proper machine to handle the trancoding and also g729 license from Digium for each call path you have active.

Does SipStation have a list of servers sending the media streams? If this information is not readily accessible and/or they are not pingable then, there is no other way for a customer to establish latency and thus the potential quality of the trunk.

I noticed audible latency during my SipStation trial both during echo tests and actual calls. My current provider sends/receives signalling on the same server/IP. This server is pingable (average 5ms) and is very responsive to qualify times (average 5-6ms) as well.

So before you go accusing me of spreading FUD why don’t you go get working on those outdated and convoluted wiki’s!

Wow you did not even read what I wrote. How could I give you a list of servers when we said it will come from the closest pop where the other end of the call is being made. We are talking hundreds and thousands of possible servers that could change anytime.

The closest POP information should be readily available upon signup. If they’re constantly changing then that’s an administrative/support nightmare on my end.

I’m sure they change for reasons such as load balancing and if servers fail but, having that info easily accessible is useful. I’m sure there aren’t hundreds of thousands of possible servers that my SipStation trunk would connect to in Toronto for example.

Nonetheless, my SipStation experience was sub par and I’m entitled to that opinion. I stated my reasons publicly so that other potential customers can make an educated guess on the service. Also, so that Sangoma support can chime in and address my concerns politely. Even if I have misunderstood some of the parameters theres no reason to chastise me if you want me to continue gradually purchasing Sangoma phones, commercial modules, and their inevitable renewal fees.

Again. You are not understanding how this works. It relates to what number you are calling or being called from and what carrier. We direct pair with lots of carriers and they driect pair with thousands of others. Not sure how I can make this any more clear. We state in SIPStation wiki media can come from anywhere and RTP ports need to be wide open. This is how real carriers handle calls. They take oath of least resistance real time to deliver a call to deliver the lowest latency on a call.

In your setup with your other carrier they are processing media so you have no clue what the true path of a call us and the latency means nothing on the trunk is all you see is the latency to their sip server. You don’t see the whole picture.

Again your tried out free trial. This is not the same as our paid service and uses totally different servers. Regardless you have a right to your opinion but you do not have a right to state facts that are not correct or misinformation.

We have thousands and thousands of happy SIPStation customers with average customer hsitoty with us of over 2 years and almost no cancelations so of course I am going to defend something you attack with wrong information. If you do not like my response feel free to go somewhere else.