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res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.

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Given the scenario where a PJSIP channel is in a native RTP bridge with direct
media and the channel is then hung up the code will currently re-INVITE the channel
back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
this greatly.

This change makes it so that if a re-INVITE transaction is in progress the BYE
is queued to occur after the completion of the transaction (be it through normal
means or a timeout).

Review: https://reviewboard.asterisk.org/r/4248/
........

Merged revisions 429409 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2014-12-12 13:06:24 +00:00
parent 8d384f3825
commit 74d43977cf
4 changed files with 44 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files

17
channels/chan_pjsip.c
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@@ -1736,22 +1736,7 @@ static int hangup(void *data)
struct ast_sip_session *session = channel->session;
int cause = h_data->cause;
if (!session->defer_terminate) {
pj_status_t status;
pjsip_tx_data *packet = NULL;
if (session->inv_session->state == PJSIP_INV_STATE_NULL) {
pjsip_inv_terminate(session->inv_session, cause ? cause : 603, PJ_TRUE);
} else if (((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS)
&& packet) {
if (packet->msg->type == PJSIP_RESPONSE_MSG) {
ast_sip_session_send_response(session, packet);
} else {
ast_sip_session_send_request(session, packet);
}
}
}
ast_sip_session_terminate(session, cause);
clear_session_and_channel(session, ast, pvt);
ao2_cleanup(channel);
ao2_cleanup(h_data);