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Hello everyone - I have the unusual challenge of making SIP VoIP work for a small business on extremely limited bandwidth... 20ish users on a 12/1 DSL line.

I'm using an RT-AC68U with 380.61. In my reading, it's my understanding that "traditional QoS" has been broken for a while, so I'm using "Adaptive QoS" with Bandwidth on auto too. I've set all my VoIP phones to highest priority using the GUI.

So here are my questions...

Leave it on Adaptive QoS?

Manually set the bandwidth or leave it on auto?

Other settings? Also, I actually have TWO of these 12/1 DSL lines so I'm debating using dual WAN but not sure if QoS and dual want can work at the same time? Any suggestions? VLAN?

Thanks in advance, I know some of the answers are already out there, but wanted to know what the latest is. Thanks.
 
I would put the VOIP on its own DSL line and have the users on the other line. 1 megabit up isn't a whole lot to work with when there's 20 people.
 
I would put the VOIP on its own DSL line and have the users on the other line. 1 megabit up isn't a whole lot to work with when there's 20 people.

Not really an easy way to do that. Remarkably using the settings above everything is working pretty well. Not sure how to add that second DSL line in other than dual WAN?


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Personally, I would use two separate routers for the two DSL lines. Connect them to a managed switch, with two separate VLANs: one for the LAN, one for VoIP.

For QoS, Adaptive QoS might work well for home, but I'm not sure I'd trust it for a business environment since it cannot be customized. Recently, I installed an RT-N66U running Toastman Tomato for a customer who needed QoS for their VoIP with 10+ users and a 25/7 Mbps DSL line.

I wish Adaptive QoS implemented some classification reports like Tomato does, to determine what traffic gets allocated to which classes. Might be doable since that info is available from Linux's tc subsystem, however since Adaptive QoS is a blackbox, it might change classes at any time without us noticing.
 
Just a follow up to say that, to my surprise, this is working really well. 25ish users with internet AND VoIP on 12/1?!

I moved the receptionist to the second DSL line and my idea was to see who is on the phone the most and maybe move that usr too, but amazingly, it's not been necessary.
 
As a point of reference

A T1/DS1 bandwidth is 1.544 Mbps - and this is 24 aLaw 64K PCM channels... G711 is 64K PCM in VOIP land, so it's roughly equal to the old school bell stuff...

So a 1Mbps broadband channel should be able to easily handle 20-25 VOIP users, esp if one uses a vocoded codec like G729/G722, and voice quality is dependent on how tight one turns the screws on the codec

VOIP does benefit from QoS, and both ethernet and wifi have QoS profiles that work best there - so worth investigating...

With most SIP based VOIP solution - keep in mind that SIP is traditionally TCP based, so that part is robust, but the voice audio path is UDP, so again, careful consideration has to be done there with regards to QoS and overall utilization in conjunction with other services...

Once you get to around 20 users on VOIP - like @RMerlin suggests, it is a good idea to consider a separate broadband connection for the VOIP gateway...
 
Once you get to around 20 users on VOIP - like @RMerlin suggests, it is a good idea to consider a separate broadband connection for the VOIP gateway...

Never was able to determine if Asus Dual Wan works with Adaptive QoS. Also if the default settings for VoIP on Adaptive QoS include UDP?

In any case it's working well for now. I only have one Asus Merlin router, so I'm just taking the most used voice users and plugging them in directly to the other dsl modem.



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