Who leads in solving VOIP issues?

Fred Marshall
Fred Marshall used Ask the Experts™
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In doing business networks, I've encountered quite a few VOIP installations.  
Many of them are provided by the ISP provider and run on the same office subnet as the computers.  Even these have troubles from time to time but are really quite simple.
Others have the VOIP set up with a separate internet connection solely for VOIP, provided by a separate VOIP provider, and are isolated from the office network with a VLAN.
In the latter case, responsibilities get blurred.
So, I'm wondering if there is a VOIP industry best practice?

Here's an example:  
One desk's phone often has a screeching noise "like grinding metal" in the audio that is audible on both ends.
The phone has been replaced a couple of times to no avail.
Other phone on the same switch end has no trouble.
I've started a Wireshark capture of the traffic.

Another example:
Calls are sometimes such that audio is only working in one direction.  So, only one participant can hear anything.

We know and understand the network.
The VOIP provider understands VOIP and their system.

What is common and best practice for dealing with things like this?
How much support and service should we expect from the VOIP provider?
Who leads in the investigation?
etc.

I'm not interested in finger-pointing, just trying to calibrate expectations.
I can well imagine that this is contract-dependent but this must be a common situation with a wealth of experiences.
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Commented:
Every job is different.  It all comes down to QOS on internal network and the edge of the Internet fabric when SIP trunks are used.  The devil is in the details so how QOS is achieved and maintained will depend on many factors that are specific to each installation. IMO there is never a need for a separate Internet connection for VOIP deployments.  Many users do this when they are flying by the seat of their pants and buy a host system off the Internets.
 https://en.wikipedia.org/wiki/Quality_of_service
Software Engineer
Distinguished Expert 2018
Commented:
Screatching sound are a sign of packet loss. (or severe latency/jitter) QOS / Routing / Bandwidth management is a big issue on this.
It can be on shared circuits great care is then needed to allow enough bandwidth for VOIP.
Bandwidth management can only effectively be done on egress, so you may need BW management in the inside (LAN) interface to limit BW.

One way traffic are NAT issues. NAT is a real big problem in VOIP setups. use 1:1 NAT as much as possible, at least disable ALG's most are implemented the wrong way, any serious VOIP provider should be able to handle FENT. (Far End Nat Traversal) it may require some fiddling with port forwards... for easier management .

Every case is specific and one can be bitten on all kinds of levels... We had a customer that could not have conversations  with clienst on a part of one mobile providers space.
Some people could not call them, they could call others...   the cause? most probably a VOIP provider that added some SBC's in their chain, possibly together with the mobile provider adding some codecs(for better sound quality)  causing UDP packet to exceed 1500 bytes.
the SIP INVITE packets never reached us.
So any working installation can easily get nasty later on.

Be sure to solve all minor issues, otherwise later on several smaller issues will cause troubles.

Author

Commented:
noci:  Thank you for the comments!
QoS is set up as follows:
All switches are Cisco SG300 series.
VOIP is running on a VLAN.
I have identified each switch port (mostly trunked) that would be carrying VOIP traffic (as well as workstation data).
I used the default Basic QoS.
I used CoS/802.1p trust mode.
I assigned default CoS level "6" to each of the ports supporting VOIP traffic.  All others remain at the default level "0".
I left the CoS/802.1p to Queue setting for "6" on queue "4".

you may need BW management in the inside (LAN) interface to limit BW.
I want to make sure that I look at this appropriately: FYI: the system is configured with a central LAN switch, that feeds two internet firewalls: One for general purpose internet connections and one strictly for VOIP service.  
If egress bandwidth is to be limited then I might assume we are concerned with *other* traffic that may interfere with the VOIP traffic.  The alternative is to limit egress bandwidth on all upstream switch ports.  Correct?  I can probably target the limits appropriately but selecting the parameters might be a bit of a challenge.  What is typical?  The general purpose internet service is 50Mbps/25Mbps.  I don't know right now the VOIP service speed but presumably that's not the target of the bandwidth limiting, right?
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If you discover the bandwidth usage of the handsets you will then know how much to allocate to the VoIP system. Most such calculations are based on the premise that not everyone will be on the phone at the same time; working out a realistic percentage of phone usage will prevent you from allocating an excessive amount of bandwidth unnecessarily and perhaps choking the office internet to death...

VoIP normally is given priority because it's so unforgiving of lost packets, regardless of cause. Having done this you then have a clear space in which to analyse other problems.
nociSoftware Engineer
Distinguished Expert 2018

Commented:
You need QoS to manage bandwidth distribution & prioritization within the LAN. (trunked ports etc.)
Alternate if you can reserve bandwidth you could set it to the max. needed for VOIP.  eff. 25Mbps.

If there is a dedicated VOIP router then that "should" be it.   Any troubles might be caused by NAT, ALG, etc. Or lack of bandwidth of aggregated VOIP traffic.

Basicly "standard" VOIP is alaw/ulaw 64kbps. (8kHz sample rate, 8 bit samples)  the gsm codec will improve on this (lower bandwith due to compressions) HD sound may take a bit more.   A reasonable estimate for bandwidth needed will be  (#concurrent connections * 64kbps.)  and take a margin for overhead like DNS, SIP, IP overhead on top of raw speed etc. say 10 concurrent connections = 1Mbps.

You can limit the egress speed of all stuff on a modem for any protocol to (bandwidth   to say 25 - 1Mbps = 24Mbps.  It would be wise to leave a gap for bursts.. so 25 - 2 = 23Mbps for all remaining traffic "should" work.
You do need to ensure the line 50/25 is effectively 25Mbps upstream... if in reality you can only get 10Mbps, the limit should be 10 - 2 = 8Mbps.!!!!  
It will make things a lot easier if the VOIP at least can get it's own public IP address. (1:1 NAT f.e., disecting traffic etc. )

Packet loss as such is not unforgiving, excessive packet loss is...  with the GSM codec you can loose quite some traffic before it becomes trouble some. a single lost packet wil not be noticable, it's a few ms. of sound. it will be a click/tick if few are missing. when the voice gets metalic you have already quite a loss of traffic depending on codecs up to 50%... (more or less evenly spread in time).

Author

Commented:
The 25Mbps up was a measured number not a quoted number....

Surely, by 1:1 NAT you don't mean 1 public address per phone do you?  So, I'm unclear as to the meaning here.

I should be clear:
The 50/25 does NOT have anything to do with the VOIP service except the traffic will compete on the LAN.  The VOIP service has a separate internet connection and it's probably sized for supporting the VOIP service only.
nociSoftware Engineer
Distinguished Expert 2018

Commented:
With 1:1 Nat i meant one IP address for an On premisses (soft)exchange.  (asterix, FusionPBX, 3CX etc.).
For Phones 1:1 nat is not realistic.   It may help to reserve RTP ports / phone on the Firewall.   (you need 2 UDP ports per conversation / phone) and allow a few for teardown of old call.
Say phone one gets 10000 - 10019, phone two: 10020-10039, ...  [ this needs to be setup on the phones as well ].. With too many phones this would become unworkable  anyway with more than a few phones an on premises a local PBX probably is better anyway. (esp. if there are a lot of "internal" conversations).   3CX, FreePBX, FusionPBX are usable systems, that can run as VM.

Phones tend to be the ones doing registration etc. Be sure that they do keep alive for NAT traversals, say have some "empty"  SIP command (Option, register etc.) every minute or so.
[ the UDP ports need to kept alive on intermediate firewalls ].

25Mbps should allow for >200 concurrent conversations if dedicated for VOIP.  On LAN ensure QoS has been setup correctly and keep all Voice traffic on a separate VLAN.

Author

Commented:
Thanks!!

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