VoIP problems and dilemma

Hi, I have VoIP softphones installed on 30 computers in an office which is located in Bulgaria. They connect to external VoIP server which is in Israel and then make calls to Turkey. My internet is not guaruanteed speed. Latency to Israel is 100ms with speedtest and now I’m installing new ISP which can give me 1gbit speeds, but not everywhere offcourse. OK, they told me the speeds to Bulgaria will be 1gbit download and upload and in most cases min 300-400mbit/s, which is very good. To Israel for example I’ll have 100mbit down and 900mbit upload. This is measured with speedtest. So, now my dilemma is how to configure speed limit using queue trees. I’ve configured such with 80mbit limit for download and 25mbit upload. Note that max upload I get is 90mbits but this is with my previous ISP. Now I must calculate it somehow for getting VoIP prioritized. Any suggestions? Thank you.

First some numbers. Depending on codec used but the best codec G.711 requires 128Kilobits per second for each VoIP phone’s conversation (i.e. 64Kbps in each direction) to ensure Quality of Service (QoS). Assuming 30 VoIP phones are used concurrently at times, a total bandwidth of 3.84 Megabits/s would be needed. Your nominal bandwidth (100Mbps) is more than enough to allow for QoS on VoIP, many times over.

However, I’d say the potential latency in your case (with the VoIP server being located far away in Israel) may be more of an issue for your VoIP quality than your Internet speed. Try pick a server in Israel and do a speedtest. That would give you a more realistic latency affecting your VoIP traffic. Latency higher than 300ms is generally unacceptable for reasonable conversations, especially for business organizations.

As for implementation of QoS on MIkroTik, it is a three step process: Classify->Mark->Prioritize. The linked article below gives an example in implementing QoS on MikroTik. You can modify to suit your VoIP.
http://networkgeekstuff.com/networking/minipost-mikrotik-qos-prioritization-example/

Ping to Israel is 100ms for each of the servers I have tried. Maybe 10-15 computers speak at the same time. I believe because SIP is not configured to be over TCP, that might be the issue.

Perhaps, the linked article by Duxtel Australia may give you much clearer info on QoS for VoIP.

https://shop.duxtel.com.au/article_info.php?articles_id=10

Good luck.
Peter F.

If ping is 100ms all the time it should not affect IP voice quality, but if it varies a lot (for example 10ms, 30ms, then 100ms) it will be a problem for voice communication.

One more thing: since you are using international SIP proxy, I would strongly recommend using encryption (which will increase each stream bandwidth by about 30-40Kbps while using standard IPsec).

I suspect other thing. My ISP maybe is limiting my total international bandwidth. Although I have between 20 and 40mbits for Israel, I’ve got 90mbits to London. They’re opening sites from Turkey, maybe somewhere there is some limitation. I’ve limited the overall speed in queue tree to 80mbit for download.

EDIT
And btw I’m seeing RTP traffic about 1kbps.

Note that if you have a Jitter buffer turned on anywhere, you’re adding latency as well (so that it can collect enough packets).

Well, pcunite’s post on QoS on VoIP is so comprehensive that I have come across on same topic. So thank you.

For reasons of security over IT, I would not use a server located in Israel. I have nothing against them though. Truth be told.

Where mikrotik has jitter buffer? Or maybe is in Bria the problem?

EDIT
OK, may network topology is Mikrotik Router RB3011 → ASUS 100mbit switch → computers . There is Domain Controller which btw I see has high CPU usage! Above 80! I read somewhere that I need
to setup Mikrotik as caching-only for DNS. I’ve tried using a guide, setup
I’ve once connected one PC directly to Mikrotik port and I saw that problems for 1 client decreased. Very interesting that was… I’m testing one group of computers - 6 PCs and I’m going to connect them also directly to the router ethernet ports.

The MikroTik does not have jitter buffer setting, your PBX, or VoIP equipment does, however. It is possible for a MikroTik to help with bytes queued that can occur at your ISP’s on-premises equipment (cable modem, etc). I’m just pointing out causes of latency.

OK, my VoIP has provided me setting for the softphones and I see that I must enable only those codecs which he gives. Also - setup explicitly some other settings, so the softphone doesn’t wonder what to use. I’ve setup almost all clients to be with cables, not Wifi. I’m going to provide another ISP sooner and make it primary. Speeds which gives are far way more better to VoIP server country and also international. Should I setup VoIP priority in my LAN. Maybe change the switch with manageable and having QoS? Should I configure VoIP priority with queues when my speed is not guaranteed? I’m going to put 2nd DC as backup and move the DHCP to Domain Controllers. Should this give better functionality?