How to use allow_contact_rewrite in pjsip in android - sip

Question

I am trying to build pjsip VoIP app for android.This is the following description given for property used in account configuration.
/**
* This option is used to update the transport address and the Contact
* header of REGISTER request. When this option is enabled, the library
* will keep track of the public IP address from the response of REGISTER
* request. Once it detects that the address has changed, it will
* unregister current Contact, update the Contact with transport address
* learned from Via header, and register a new Contact to the registrar.
* This will also update the public name of UDP transport if STUN is
* configured.
*
* See also contact_rewrite_method field.
*
* Default: 1 (yes)
*/
pj_bool_t allow_contact_rewrite;
Until now via address is only changed by pjsip.how exactly does this property work??i can see no differences??

Answer 1

This option only kick in once the pjsip client is behind some sort of NAT.
i.e.
[pjsip client / e.g. 10.0.100] -> [NAT / e.g. 1.2.3.4] -> [SIP Server / e.g. 2.3.4.5]
When PJSIP registers with the sip server with the "contact" address of 10.0.100:5060 the SIP Server will send back a response saying that it came from 1.2.3.4: (because of the NAT).
This is when the allow_contact_rewrite kicks in and unregisters the registration and re-registers it with the 1.2.3.4: contact address.
Why it does this is so that if the sip proxy server needs to contact/send request to the SIP UAS it uses the contact header address.
This feature is "assuming" many that the NAT support hole punching to keep the contact address valid over a range of time.
Another option is to setup STUN so that PJSIP can figure out this information ahead of time so that it "knows" the external ip address before it registers (although there is symmetric NAT which can make STUN useless).
All of these options are to try get a setup so that the client can be contacted through the internet from the SIP Proxy server to the pjsip client when a "event" happens (e.g. phone call).
Personally I've found none of these setups to get a contact address that is connectable by the SIP Porxy server to work at all.
The best setup when coming from the internet client that I use is to always use TCP only (no UDP). Enable rport and TCP keep-alive in pjsip. As long as the sip server respects rport and the tcp/ip connection stays alive then everything works fine (i.e. the sip proxy can send requests down the existing open TCP socket).
That just leaves the problem of media connections between the two endpoints, which you have to rely of ICE/STUN/TURN/NAT hole punching to get that going.