![]() I have disabled SIP ALG but I'm still experiencing problems. Disallows 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.Writing incorrect port values greater than 65536 is also common in many of these routers. ![]() missed semi-colon " " in header parameters). Many SIP ALG routers corrupt the SIP message when writing into it (i.e. 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). 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. ![]() A SIP ALG router rewrites the REGISTER request to the proxy doesn't detect the NAT and doesn't maintain the keepalive (so incoming calls will be not possible). ![]() Many SIP proxies maintain the UDP keepalive by sending OPTIONS or NOTIFY messages to the UA, but they just do it when the UA has been detected as NAT'd during the registration. Common routers just maintain the UDP "connection" open for a while (30-60 seconds) so after that time the port forwarding is ended and incoming packets are discarded by the router. 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). Lack of incoming calls: When a UA is switched on it sends a REGISTER request to the proxy in order to be localisable and receive any incoming calls.Therefore if you are experiencing problems we recommend that you check your router settings and turn SIP ALG off if it is enabled. This can give you unexpected behaviour, such as phones not registering and incoming calls failing. SIP ALG modifies SIP packets in unexpected ways, corrupting them and making them unreadable. How can it affect VoIP?Įven though SIP ALG is intended to assist users who have phones on private IP addresses (Class C .X), in many cases it is implemented poorly and actually causes more problems than it solves. A NAT router with a built-in SIP ALG can re-write information within the SIP messages (SIP headers and SDP body) making signalling and audio traffic between the client behind NAT and the SIP endpoint possible. If the SIP proxy doesn't provide a server-side NAT solution, then an ALG solution could have a place.Īn ALG understands the protocol used by the specific applications that it supports (in this case SIP) and does a protocol packet-inspection of traffic through it. In some scenarios, some client-side solutions are not valid, for example, STUN with symmetrical NAT router. Generally speaking, ALG works typically in the client side LAN router or gateway. There are various solutions for SIP clients behind NAT, some of them in the client side (STUN, TURN, ICE), others are in the server side (Proxy RTP as RtpProxy, MediaProxy). Many routers have SIP ALG turned on by default. Its purpose is to prevent some of the problems caused by router firewalls by inspecting VoIP traffic (packets) and if necessary modifying it. Find more details how to build an app with more features and preferable design at ALG stands for Application Layer Gateway and is common in all many commercial routers. This app is only demonstration of what you can achieve with ABTO VoIP SIP SDK for Android. Enable/disable video, mic, and speaker during the call.Make voice and video calls to most sip clients with or without registration on SIP server (see list of other products on for details).It is super easy to use and has a lot of benefits for end users. The application represents most of available functionality of the SDK, but calls are terminated after 1 minute. ABTO VoIP SIP Softphone is based on ABTO VoIP SIP SDK for Android that allows development of custom sip softphones.
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