wiki:Symbian_AP_Reconnection

Version 4 (modified by bennylp, 10 years ago) (diff)

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Symbian Access Point Reconnection

Table of Contents

  1. Problem description
  2. Issues
    1. Restart everything
    2. Selective update
  3. Scenarios
    1. Access point disconnection

This article describes some issues and their corresponding solutions related to access point disconnection, reconnection, and change in your Symbian PJSIP applications.


Problem description

Access point disconnection and reconnection are scenarios that need to be handled in mobile applications. Few issues or scenarios related to this for example are:

  • user moves outside the range of a Wi-Fi access point (AP) and lost the connection
  • user moves outside the range of one AP and reconnect to another
  • the handset may get new IP address if user reconnects to different AP

Each of the scenarios above may need different handling in the application.


Issues

When the connection is reconnected, the handset may get different IP address than what it previously got. There are few ramifications of this, for example if PJSUA-LIB is used:

  • the SIP registration needs to be updated with a new Contact URI
  • the account URI also needs to be updated
  • media addresses need to be updated
  • if there is ongoing dialog, the remote party needs to be informed with new Contact URI as well as new media (RTP/RTCP) addresses.

Note that at this time of writing (2009/02/25, PJSIP v1.1), PJSIP does not have the capability to monitor the status of the underlying IP connection used by the application, so the application must implement the detection itself (for example using connection progress monitor in Symbian). Having said that, PJSIP does have some capability to detect some IP address change scenarios, for example by monitoring the IP address in SIP REGISTER response or in STUN Binding response when ICE transport is used and STUN is enabled.

Once the application has detected that the IP interface address has changed, there are two solutions to inform PJSIP about this.


Restart everything

The most straightforward solution is of course to restart everything, which means in pjsua terms to call pjsua_destroy() and followed by pjsua_create(), pjsua_init(), and so on. While this solution may sound crude, it is the easiest to do and as will be explained later it is not considerably worse then the more refined alternative.


Selective update

Alternatively there may be a way to allow the stack to continue to run, updating the address information when necessary. This approach will require some specific features to be used, as well as some actions by the application when it detects that the IP address has changed.

The specific configuration and tasks will be explained below.

Account Contact URI update

Task:
The account URI needs to be updated with the new address, and re-registration is necessary to inform the registrar about the new URI.
Solution:
PJSUA-LIB has the capability to detect the (SIP) IP address change based on the response of REGISTER request and automatically update the registration with the correct IP if it detects that the IP/port seen by the server is different than the address specified in the Contact URI. This feature is enabled by default, via the pjsua_acc_config.allow_contact_rewrite setting.

So the solution is simply to trigger the re-registration by calling pjsua_acc_set_registration() function (after the new connection is up of course). The PJSUA-LIB will send re-REGISTER request, check the IP address/port in the response, and re-REGISTER again and update the account URI as necessary.

Media addresses update

Task:
The media (RTP/RTCP) addresses in PJSUA-LIB are normally determined during PJSUA-LIB startup, hence they need to be updated with the new address.
Solution:
If ICE media transport is used, and STUN is enabled on the media transport, then the media transport will automatically update its publicly mapped IP address from the STUN Binding response. The transport should send STUN Binding request periodically (approximately every 15 seconds) as NAT keep-alive mechanism, so the address change will be detected by the transport automatically during this operation.

Note that at present there is no API to explicitly request the ICE media transport to initiate STUN Binding request immediately.

Call in progress issues

Task:
Dialog's Contact URI needs to be updated.
Solution:
The dialog's Contact URI is set initially when the dialog is created, from the account's Contact URI. While at the PJSIP level the pjsip_inv_reinvite() allows changing of Contact URI via the new_contact argument, currently this feature is not used by PJSUA-LIB, i.e. the pjsua_call_reinvite() does not allow the application to change the Contact URI.

So this is an open issue.

Changing of RTP/RTCP media addresses:

Scenarios

Access point disconnection