Turning Off SIP ALG or SIP Transformations

First a little background on SIP ALG (Application Layer Gateway). It is a security component of a router or NAT that allows VoIP traffic to pass through from the private to the public and vise a versa through the firewall when NAT and NAPT is being used. The SIP ALG inspects the traffic and then repackages it as it allows it to continue on to its destination. Many routers incorporate this protocol, however unfortunately after the inspection some of the information is lost or rearranged which results in the return RTP not making it to the final destination inside the LAN. The result is one-way audio where the caller can hear the person inside the network behind the router, but that user cannot hear the person on the outside. The RTP data never makes it back to him.

SIP ALG on a router does the following:

Controls SIP (Session Initiation Protocol)activity by limiting the duration with inactivity media timeout features.
It protects the SIP proxy server from a DoS (Denial of Service) attacks from outside the network.
It allows passage of data when the session occurs while using NAT (Network Address Translation) mode and special routing modes.

What are SIP Transformations?

This setting is found primarily on a SonicWall and are basically SIP ALG and is well known to cause issues with some VoIP services. These issues can be one-way audio, phone registration issues, or even dropped calls. Most VoIP carriers will request the user to disable (or uncheck Enable) for SIP Transformations and enable consistent NAT (Network Address Translation) on a SonicWall if these issues are occurring.

VoIP Issues? Try disabling SIP ALG

Most routers have SIP ALG enabled by default. Disabling this setting is easy and should be attempted as a way to troubleshoot and resolve some VoIP issues including one-way audio. Below are two screenshots showing where on a Linksys WRT1900AC and on a SonicWall they can be found. There are also screenshots of where to disable SIP ALG on a Netgear AC1900 and a version of Tomato firmware.

Linksys WRT1900 ALG setting.

Disabling SIP ALG on a WRT1900 Linksyst router.

Most good routers have advanced routing rules that can help accomplish many obstacles, such as Port Forwarding and other advanced routing features.

Sonic Wall screenshot showing the SIP Transformation setting.

Screenshot of Sonicwall SIP Transormations.

As shown, the Sonicwall includes a setting to Enable consistent NAT (Network Address Translation). This setting can help ensure that the NAT address remains the same, helping voice traffic (RTP) to find the correct endpoint.

The Netwgear AC1900 under WAN settings.

SIP ALG on a Netwgear AC1900 router.

The Netgear Nighthawk series of routers are fast and can handle numerous devices and are good for networks that use VoIP. The AC1900 offers excellent value and performance.

Tomato software go to the NAT Helpers screen.

The SIP ALG setting on Tomato firmware.

Depending on the version of Tomato, SIP ALG cab be found under Advanced then Conntrack/Netfilter in the Tracking/NAT Helpers section. If you find SIP checked then SIP ALG is enabled. Uncheck it to disable it.

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Enabling Constant NAT

Sonicwall firewall devices have a setting of Constant NAT. Some VoIP providers will suggest enabling constant NAT along with disabling SIP Transformations. We suggest that you follow their suggestion.

Changing outbound port numbers, which are done by routers and/or Sonicwall devices, will cause issues with the VoIP traffic which can result in one-way audio and dropped calls. Keeping the same NAT pinhole assignment allows RTP traffic to complete its travels to the correct destination. Constant NAT helps with RTP traveling through the firewall and using this setting as a troubleshooting tool can help resolve many one-way audio issues with VoIP.

SIP (Session Initiation Protocol)

SIP is an application layer control protocol that supports features that are needed when initiating and ending communication. SIP does not provide services itself, therefore it works in conjunction with other protocols to provide those services. One such service is RTP which carries the voice for a VoIP call.

SIP (Session Initiation Protocol)

The five parts of setting up and terminating calls that SIP handles are:

User Location- Determines where the end system is that will be used for a call.
User Availability: Determines of the availability of the called user to engage in a call.
User Capabilities: Determines of the media and parameters which will be used for the call.
Session Setup: Establishes the session parameters from both parties (ringing, etc.).
Session Management: Starts the services including transfer, termination, and modifying the sessions parameters.

SIP Response Codes

Should you have access to a SIP request and call you can use the response codes to gain valuable troubleshooting information.

1xx = informational responses

100 Trying
180 Ringing
181 Call Is Being Forwarded
182 Queued
183 Session Progress

2xx = success responses

200 OK
202 accepted: Used for referrals

3xx = redirection responses

300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
380 Alternative Service

4xx = request failures

400 Bad Request
401 Unauthorized: The request requires user authentication. Issued by registrars and UASs. (407 - Proxy Authentication Required is used by proxy servers.)
should use proxy authorization 407
402 Payment Required (Reserved for future use)
403 Forbidden
404 Not Found: User not found
405 Method Not Allowed
406 Not Acceptable
407 Proxy Authentication Required
408 Request Timeout: Couldn't find the user in time
410 Gone: The user existed at one time, but is no longer available at the server and no forwarding address is known.
413 Request Entity Too Large
414 Request-URI Too Long
415 Unsupported Media Type
416 Unsupported URI Scheme
420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server
421 Extension Required
423 Interval Too Brief
480 Temporarily Unavailable
481 Call/Transaction Does Not Exist
482 Loop Detected
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
491 Request Pending
493 Undecipherable: Could not decrypt S/MIME body part

VoIP Router Settings