SIP Phones or IP PBX Behind an AOS Device

[Anonymous]

I am starting to put some Polycom and misc other sip phones behind the NV4430.  We do not use a NV7000 series PBX, as we connect to a Genband C1500 directly and use centrex features. I got them working great behind the NV4430 by disabling the SIP ALG and enabling Main SIP service.

My question is now: Is there any advantage of using the SIP proxy for my application? Is this just for inner office communications if the Internet was to go down?

sip_

Use this page to enable/disable SIP Proxy services and modify SIP Proxy parameters.

SIP Proxy SIP Proxy Server: Enabled Transparent Mode: Enabled Proxy Dial-String Source: Proxy Domain:

SIP Proxy Grammar 'From' Field Host: 'Request-URI' Field Host: 'To' Field Host:

Allowed Remote SIP Proxy Servers SIP Proxy can communicate with any SIP server. Add an 'Allowed Remote Sip Server' to restrict.

Alternative Remote SIP Proxy Servers There are no remote proxy servers configured.

[Anonymous]

:

Thank you for asking this question in the support community.  Yes, there are many more benefits to using the SIP Proxy and SIP Proxy User Templates, other than survivability (as you mentioned).  For detailed information please review the Configuring SIP Proxy in AOS and Configuring Proxy User Templates in AOS documents, but I have added an general overview below.

Using the SIP Proxy feature allows AOS products to forward SIP messages between endpoints (phones and softswitches) without being required to interpret the messages as is done with back-to-back user agent (B2BUA) devices.  When using the AOS SIP Proxy feature, customers are not restricted to using ADTRAN-compatible (B2BUA) phone models. Unlike the B2BUA, the SIP Proxy is not feature dependent. As a result, adding new phone or softswitch features does not require corresponding features in the AOS product, or additional configuration to support customer endpoints. Configuration is simplified because creating voice users, voice trunks, and dial plans is not required on the AOS product. In addition to increased interoperability, the SIP Proxy feature is switchboard independent and increases survivability options in the AOS products.

The user template makes it possible to route traffic for users not already known by the Session Initiation Protocol (SIP) Proxy. This feature can also direct certain calls to local external public switched telephone network (PSTN) gateways. Users are dynamically added to the proxy user database when an INVITE or SUBSCRIBE message is received from the user, allowing SIP messages to be routed for the identified user.

As a general guideline, SIP Proxy should be used when phone traffic is routed through the ADTRAN device, and SIP Proxy User Templates should be used when endpoints don't REGISTER, such as a phone system.

I hope that makes sense, but please do not hesitate to reply to this post with any additional questions or information.  I will be happy to help in any way I can.

Levi

[Anonymous]

Thank you for the info and documents, that was very helpful. I am just not sure which one to use (for my application) of the three types of proxy: outbound, stateful, and transparent.

[Anonymous · Accepted solution]

:

Here are some guidelines:

• If there is a phone system behind the ADTRAN (such as an IP PBX):
  • Configure proxy User Templates
• If the phones are sending their SIP information to the public IP address of the SIP server and do not know the ADTRAN unit exists (most common application😞
  • Configure proxy Transparent
• If the phones are sending their SIP information to the ADTRAN unit's IP address as the destination:
  • Configure proxy Stateful

The Transparent Proxy mode is best for new installations when the phone does not need to be aware of the device presence (transparency). In this mode, the softswitch is configured as the server and the registrar (and Outbound Proxy when required). The phones are not aware of the AOS device, and phones are not required to use the same softswitch. Requests sent from the phones are routed to the destination IP address of the softswitch, and the AOS device does not require configuration to be aware of the softswitch. In Transparent Proxy mode, the softswitch address configured in the phone is the external softswitch, and SIP signaling is sent to the Layer 3 destination address of the external softswitch.

The Stateful Proxy mode is best suited for existing installations due to the ease of converting from B2BUA to IP phones without modifying the phone configurations. In this mode, phones or user agents (UAs) rely on the SIP server to act as the proxy, and the configuration is very similar to the existing B2BUA. The AOS product takes on the role of proxy. The UA is not aware of other SIP servers on the network in this mode. Each phone is configured to register to the proxy, and the phone is not aware of the softswitch(es). Either the softswitch(es) host name(s) or IP address(es) must be programmed in the system, enabling communication between the two. All phones must use the same softswitch or set of softswitches, and the phone must send all requests to the destination IP address of the AOS device. In Stateful Proxy mode, the softswitch address configured in the phone is the proxy (AOS device), and SIP signaling is sent to the Layer 3 destination address of the proxy (AOS device).

Levi

[Anonymous]

:

I went ahead and flagged this post as “Assumed Answered.”  If any of the responses on this thread assisted you, please mark them as either Correct or Helpful answers with the applicable buttons.  This will make them visible and help other members of the community find solutions more easily.  If you still need assistance, I would be more than happy to continue working with you on this - just let me know in a reply.

Levi