Se o SIP-ALG é um problema com os telefones VoIP, por que ele está sempre ativado por padrão ou ainda sozinho, usado como protocolo?
O SIP-ALG permite que um roteador NAT reescreva as informações dentro das mensagens SIP (cabeçalhos SIP e corpo SDP).
A intenção é evitar alguns dos problemas causados pelos firewalls do roteador, inspecionando o tráfego de VoIP (pacotes) e, se necessário, modificando-o.
Isso faz sinalização e tráfego de áudio entre o cliente por trás do NAT (um telefone VoIP, por exemplo) e o ponto de extremidade SIP possível.
Sem SIP-ALG você pode ter áudio unidirecional em conversas VoIP (veja abaixo).
Infelizmente, o SIP-ALG em muitos roteadores é mal implementado e causa mais problemas do que resolve.
Se você tiver um desses roteadores problemáticos, recomenda-se que o SIP-ALG seja desativado e outros métodos sejam usados para garantir que o VoIP funcione corretamente.
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.
Fonte O que é o SIP ALG e por que a Gradwell recomenda que eu o desligue?
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.
Fonte Roteadores SIP ALG
Por que uma maneira de áudio ocorre?
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 differentiate them.
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 process).
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.
Fonte Por que eu recebo áudio de uma forma ao fazer chamadas telefônicas?