Thank you for asking this question in the support community. Is this the guide you are looking for is the Configuring SIP Proxy in AOS? Also, this post may be beneficial when deciding which type of proxy to use: SIP Phones or IP PBX Behind an AOS Device
Please, do not hesitate to reply with any additional questions.
Unfortunately, the information does not answer my question. I was looking for actual detail, which I believe I read from an Adtran document. Specifically, I am looking for packet manipulation detail.
1. How does the transparent proxy manipulate the SIP UDP packets? Does it change the IP address and UDP packet transparently, i.e., without knowledge of the SIP client?
2. How does the SIP outbound proxy manipulate the SIP UDP packet? Does the same thing as transparent,except with the knowledge of the SIP client to provide QOS functionality and possibly SIP forking? Does it do anything with the SIP message?
3. How does the SIP stateful proxy manipulate the SIP UDP packet? Does it go beyond what the outbound proxy does and manipulates the contact information in the SIP message? Is there anything else in the SIP message that it might manipulate?
Just looking for the exact detail regarding IP/UDP/SIP manipulation in each of the proxy types. Appreciate your help and responses.
I too would like answers to these three questions.
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Regardless of the SIP proxy mode, the 3 main components of the SIP Proxy are as follows:
a. To route SIP Requests to the proper destination and route SIP responses back to the originator, without having to understand the purpose of the message
b. To create dynamic holes in the AOS Firewall for the RTP Media
c. To provide rollover and failover (survivability) in the case of a softswitch failure
When SIP requests are processed by the SIP proxy, the ADTRAN will perform the necessary translation of private IPs used in the SIP header (Via, Contact, etc) and SDP before forwarding the request. For example, if the contact header in a received SIP request is firstname.lastname@example.org, the SIP proxy would change the host portion of the contact header to the NAT’d address configured on the firewall (i.e. email@example.com). This same functionality would occur on all three proxy modes.
The SIP proxy mode that you need to use depends mainly on the SIP phone’s configuration. For transparent mode, the SIP phones are configured to communicate directly with the soft switch. The phones are not aware of the ADTRAN. This is the mode we typically recommend for hosted PBX applications. For stateful mode, the SIP phones are configured with the ADTRAN as their SIP server. They are not aware of the soft switch. This is the mode we typically recommend for existing IP PBX or SIP phones that are already configured to communicate with existing Back-to-Back-User-Agent (B2BUA) installations. However, for SIP Trunking between an IP PBX and a provider, we would recommend using the ADTRAN’s B2BUA rather than the SIP proxy. This is available in the ADTRAN IP Business Gateways and routers with the SBC feature pack.
For outbound mode, the phones are configured with the softswitch as the server and registrar, but with the ADTRAN device as the outbound proxy.
Finally, I found the document I requested in my original question. It is labeled "Configuring IP Interfaces for SIP in AOS IPBGs." It provides the details on how each proxy type manipulates the SIP message and packet.
This question can be considered answered. Appreciate the responses.