pjsip: replace 180 by 183 if SDP negotiation has completed

The caller endpoint hears dead silence if a callee replies 180 (without SDP)
and the caller already received 183 (with SDP).
It happens because Asterisk sends 180 (WITH SDP) to the caller,
there are not incoming RTP packets from the callee
and Asterisk does not generate inband ringing,
so there are not any outgoing RTP packets to the caller.

This patch replaces 180 by 183 if SDP negotiation has completed,
as if the caller endpoint is configured with "inband_progress=yes".

In this case Asterisk will generate inband ringing untill Asterisk receive
incoming RTP packets from the callee.

ASTERISK-27994 #close

Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 9c27f33b11d5f5993139927de387e626a229e029..8508631296a44b25615ddba69b0f670f6ffa69f4 100644 (file)
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1592,7 +1592,9 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
        switch (condition) {
        case AST_CONTROL_RINGING:
                if (ast_channel_state(ast) == AST_STATE_RING) {
-                       if (channel->session->endpoint->inband_progress) {
+                       if (channel->session->endpoint->inband_progress ||
+                               (channel->session->inv_session && channel->session->inv_session->neg &&
+                               pjmedia_sdp_neg_get_state(channel->session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE)) {
                                response_code = 183;
                                res = -1;
                        } else {