If SIP-ALG is such an issue with VoIP phones, why is it always enabled by default or yet alone even used as a protocol?
SIP-ALG allows a NAT router to re-write information within the SIP messages (SIP headers and SDP body).
The intention is to prevent some of the problems caused by router firewalls by inspecting VoIP traffic (packets) and if necessary modifying it.
This make signaling and audio traffic between the client behind NAT (a VoIP phone for example) and the SIP endpoint possible.
Without SIP-ALG you may have one-way audio on VoIP conversations (see below).
Unfortunately the SIP-ALG in many routers is poorly implemented and causes more problems than it solves.
If you have one of these problem routers it is recommended that the SIP-ALG is disabled and other methods used to ensure VoIP works correctly.
Gradwell and other providers often find that the SIP ALG modifies SIP packets in unexpected ways, corrupting them and making them unreadable. This can give you unexpected behaviour, such as phones not registering and incoming calls failing.
Source What is SIP ALG and why does Gradwell recommend that I turn it off?
SIP ALG problems
The main problem is the poor implementation at SIP protocol level of
most commercial routers and the fact that this technology is just
useful for outgoing calls, but not for incoming calls:
- Lack of incoming calls: When a UA is switched on it sends a REGISTER to the proxy in order to be localizable and receive incoming
calls. This REGISTER is modified by the ALG feature (if not the user
wouldn't be reachable by the proxy since it indicated a private IP in
REGISTER "Contact" header). Common routers just mantain the UDP
"conntection" open for a while (30-60 seconds) so after that time the
port forwarding is ended and incoming packets are discarded by the
router. Many SIP proxies mantain the UDP keepalive by sending OPTIONS
or NOTIFY messages to the UA, but they just do it when the UA has been
detected as natted during the registration. A SIP ALG router rewrites
the REGISTER request so the proxy doesn't detect the NAT and doesn't
mantain the keepalive (so incoming calls will be not possible).
- Breaking SIP signalling: Many of the actual common routers with inbuilt SIP ALG modify SIP headers and the SDP body incorrectly,
breaking SIP and making communication just impossible. Some of them do
a whole replacing by searching a private address in all SIP headers
and body and replacing them with the router public mapped address (for
example, replacing the private address if it appears in "Call-ID"
header, which makes no sense at all). Many SIP ALG routers corrupt the
SIP message when writting into it (i.e. missed semi-colon ";" in
header parameters). Writting incorrect port values greater than 65536
is also common in many of these routers.
- Dissallows server side solutions: Even if you don't need a client side NAT solution (your SIP proxy gives you a server NAT solution), if
your router has SIP ALG enabled that breaks SIP signalling, it will
make communication with your proxy impossible.
Source Routers SIP ALG
Why does one way audio occur?
The most common cause of one way audio is routers, because many
routers are not built with VoIP in mind. Most ISPs will give you one
IP, which is the address you use on the internet and allows other
computers to find you.
When you connect devices and computers to your router, it has to share
this IP out amongst all of your devices. Therefore it uses something
called Network Address Translation (NAT). Using this, each of your
devices get their own internal IP address so that your router can
This isn’t used for talking to the internet, and is an internal
address only. When your computer or other devices communicate with the
internet the IP address that your internet service provider has given
you is used (please see this article for more information on this
This generally isn’t a problem when performing everyday tasks on the
internet, as you mostly retrieve information, and not have it sent
directly to your computer.
However, a phone call is a two way process. You send audio out via the
internet and the other party sends theirs back to you. This causes a
dilemma, as the audio will have your public IP address on it, but not
the internal address of your device. Your router receives the
information, but doesn’t know what to do with it. Therefore it ignores
it. This means that the other person hears you talking, but no matter
what they say you don’t hear them. Their part of the conversation
simply isn’t getting through.
Source Why do I get one way audio when making phone calls?