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Fix One Way VoIP Audio (SIP, NAT and STUN)

The Problem - When making VoIP calls (particularly with SIP) you can ring phone numbers but once the call is answered there is either no voice or it is only one way.

The Cause - I am pretty sure the cause of this will be the same regardless of what protocol you are going to use for your VoIP solution but I only have experience of SIP. So this will definitely be an issue with SIP but I haven't confirmed it with the other protocols.

The problem arises because VoIP uses dynamic UDP ports for each call. This causes problems when traversing a NAT device for two reasons; the NAT device changes the source port of outbound packets as part of the NAT process. The second is because UDP by its very nature is designed for one way traffic (broadcasts, video stream etc). Where TCP traffic is bi-directional across the one connection UDP can have 1 connection for inbound and another for outbound meaning they can use different ports. If the inbound connection uses different ports as the outbound connection the inbound traffic will be dropped because the NAT device does not have a mapping for it in its NAT table. If you are confused by now I suggest you read up on NAT first.

What is SIP and why is it important to VoIP Just as TCP/IP is not a protocol by itself but rather a family of protocols like TCP, IP, PPP, PPTP, ARP etc so is VoIP. There are several protocols you can use with VoIP each having their own pros and cons. The one we will focus in this article though is SIP. SIP stands for Session Initiate Protocol. It is responsible for setting up the call, ringing, signalling, engaged tones etc.

In most SIP environments there will be several VoIP calls in use concurrently. Every one of these calls will be managed through the VoIP switch, each one requiring its own voice channel. Each channel (or phone call to look at it another way) must use a unique port. If there are 100 concurrent VoIP calls in use there must be 100 ports available for the VoIP switch to allocate to each call. This is where SIP comes in. It basically controls everything that is needed in setting up the call. For each call SIP will find a spare port, allocate it, send these details to all parties, set the call up and ring the phones. Once the call has finished SIP terminates the session and informs the phone switch that this port can be reassigned to another call.My site 로투스홀짝

The range of ports is usually configurable, Avaya for example allow you to configure this in the VoIP portion of the system config. The default range for Avaya VoIP is 49152 to 53246. This gives us a possibility of 4094 concurrent VoIP calls licensing permitting.

In a LAN environment this is not a problem as firewalls usually permit all traffic on all ports for all devices. Once the internet is involved where the traffic has to traverse a NAT and firewall we start to run into problems. In the Avaya example above it can pick a port anywhere in the range of 49152 to 53246. You can't just open this port range to the internet. A range of 4000 ports open isn't very secure.

How SIP is meant to work on the internet As with all network traffic one endpoint must initiate the connection first. This means at least one port must be open using port forwarding to the VoIP switch. SIP usually runs on port 5060. For the two offices to call each other both sites must have this port being forwarded to the phone switch. When you read documentation on SIP most of it will say that this is all you need to do...But in all likelihood this is not the case.

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VoIP Business Phones Need to SIP It!

What the heck IS a SIP trunk anyway?

OK, SIP trunks are basically just virtual phone lines, plain and simple.

They only work with an IP enabled PBX in a business environment, so aren't used in residential applications (unless you gotta LOT of kids on the phone)

So SIP is Session Initiated Protocol, this is basically Geek for a "computer language that carries your voice over an internet or MPLS connection to make a phone call". It's not that complicated once you understand what you need to use SIP VoIP:

1. You need an IP PBX, you cannot use SIP with other VoIP devices.

2. You need to have a quality Internet connection, or perhaps an MPLS WAN set up for branch to branch calling if this is your need.

3. You must have a quality SIP provider, because we get what we pay for, and we don't want to sound like Yoda or Yogi Bear.

When do we use SIP Trunking?

The main business drivers for SIP VoIP is the same as any larger scale VoIP deployment - to save money.

First off, let's not forget the fundamental way VoIP saves us money is CONVERGING our voice and internet data on one pipe, generally a dedicated T1 or larger, mission critical type circuit. Then we get to fire the phone company and they can take those old expensive lines (or PRIs) with them.

By using SIP trunks, we can many times reduce the number of "talk paths" we need coming into our facility compared to "old phone technology", thus cutting costs even further.

A by product of almost any VoIP deployment is "dynamic bandwidth allocation" this means we're delivering the VoIP over our internet pipe, and when we're not talking on the phone, the entire internet pipe is available for our use.

Additional economies of scale occur when we realize the MORE Branches we have, the MORE we can consolidate the required lines, many times bringing all the lines (SIP trunks) into a centralized location, and then firing the phone company and all their expensive lines at each location.

A good example of this is a regional bank. Ever noticed how you always call just one number, but you can get to any of the eight branches? That's because they're using SIP trunks, all the branches are connected together using either an internet VPN or an MPLS WAN and everyone is an extension off this one PBX back at the main branch. My site 로투스홀짝

How do we buy SIP trunks?

There are many flavors and permutations. Basically the wisest thing to do is have "one throat to choke" when it comes to this stuff.

This means, try and find a provider that will deliver a reasonably priced dedicate internet connection, and the SIP trunks all in one package. This way if anything ever doesn't work, you make one call.

Of course, you could also have an issue with your "phone guy" and the IP PBX itself, but it's much more likely to be the VoIP or internet service itself.

Other options are using a Telecom Broker. These folks really know the ins and outs (or they should) of VoIP and SIP in general, and also specific providers in your area.

Telecom Brokers shop the market, so you don't have too, and in conjunction with your PBX distributor should be able to fashion a great solution that both saves money and is every bit as reliable as your existing "old" phone service, just less expensive and more efficient!

How does SIP pricing work?

Well, again, lots of options here, and generally you get what you pay for. Most good providers will be unlimited INBOUND minutes, and then metered OUTBOUND calling. We do away with long distance charges, because every call is counted against our minutes.

Most providers will bundle say 500 outbound minutes on each SIP trunk, and then aggregate the minutes across the entire platform. So 10 SIP trunks will have 5,000 outbound minutes - call across the street or across the country.

This equates to like 83 hours OUTBOUND a month, so even if you had 20 people at the business, they would each have to be making a solid 30 minutes of outbound calls a day to use all this SIP, probably not gonna happen, unless you're a call center - and they have different pricing and needs.

Even so, if you make a lot of long distance calls, then SIP will be cheaper per "line" and also per minute anyway, shop it around, and don't forget the INBOUND is always free anyway.

Long and the short of it, don't be too worried that it's not "unlimited" outbound calling, it's virtually unlimited.

Some SIP providers call it unlimited anyway, and then tag you if you are a call center; it's in their agreement, in the fine print. Again, use a broker to determine what you need and who's the best provider for you.