diff options
| -rw-r--r-- | channels/chan_h323.c | 2 | ||||
| -rw-r--r-- | channels/chan_mgcp.c | 2 | ||||
| -rw-r--r-- | channels/chan_sip.c | 10 | ||||
| -rw-r--r-- | channels/chan_skinny.c | 2 | ||||
| -rw-r--r-- | include/asterisk/rtp.h | 2 | ||||
| -rw-r--r-- | main/rtp.c | 2 |
6 files changed, 10 insertions, 10 deletions
diff --git a/channels/chan_h323.c b/channels/chan_h323.c index 5734fbb58..720236cb9 100644 --- a/channels/chan_h323.c +++ b/channels/chan_h323.c @@ -914,7 +914,7 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data res = 0; break; case AST_CONTROL_SRCUPDATE: - ast_rtp_update_source(pvt->rtp); + ast_rtp_new_source(pvt->rtp); res = 0; break; case AST_CONTROL_SRCCHANGE: diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c index 319ce9101..787d45d16 100644 --- a/channels/chan_mgcp.c +++ b/channels/chan_mgcp.c @@ -1454,7 +1454,7 @@ static int mgcp_indicate(struct ast_channel *ast, int ind, const void *data, siz ast_moh_stop(ast); break; case AST_CONTROL_SRCUPDATE: - ast_rtp_update_source(sub->rtp); + ast_rtp_new_source(sub->rtp); break; case AST_CONTROL_SRCCHANGE: ast_rtp_change_source(sub->rtp); diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d4ff2f788..31c910a88 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -6188,7 +6188,7 @@ static int sip_answer(struct ast_channel *ast) ast_setstate(ast, AST_STATE_UP); ast_debug(1, "SIP answering channel: %s\n", ast->name); - ast_rtp_update_source(p->rtp); + ast_rtp_new_source(p->rtp); res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE); ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED); } @@ -6223,7 +6223,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame) if ((ast->_state != AST_STATE_UP) && !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { - ast_rtp_update_source(p->rtp); + ast_rtp_new_source(p->rtp); if (!global_prematuremediafilter) { p->invitestate = INV_EARLY_MEDIA; transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE); @@ -6546,11 +6546,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data res = -1; break; case AST_CONTROL_HOLD: - ast_rtp_update_source(p->rtp); + ast_rtp_new_source(p->rtp); ast_moh_start(ast, data, p->mohinterpret); break; case AST_CONTROL_UNHOLD: - ast_rtp_update_source(p->rtp); + ast_rtp_new_source(p->rtp); ast_moh_stop(ast); break; case AST_CONTROL_VIDUPDATE: /* Request a video frame update */ @@ -6569,7 +6569,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data } break; case AST_CONTROL_SRCUPDATE: - ast_rtp_update_source(p->rtp); + ast_rtp_new_source(p->rtp); break; case AST_CONTROL_SRCCHANGE: ast_rtp_change_source(p->rtp); diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c index c7341376a..dd0f7466d 100644 --- a/channels/chan_skinny.c +++ b/channels/chan_skinny.c @@ -4256,7 +4256,7 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s case AST_CONTROL_PROCEEDING: break; case AST_CONTROL_SRCUPDATE: - ast_rtp_update_source(sub->rtp); + ast_rtp_new_source(sub->rtp); break; case AST_CONTROL_SRCCHANGE: ast_rtp_change_source(sub->rtp); diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index e7c80c58a..aa38ee0fa 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -217,7 +217,7 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level); int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc); /*! \brief Indicate that we need to set the marker bit */ -void ast_rtp_update_source(struct ast_rtp *rtp); +void ast_rtp_new_source(struct ast_rtp *rtp); /*! \brief Indicate that we need to set the marker bit and change the ssrc */ void ast_rtp_change_source(struct ast_rtp *rtp); diff --git a/main/rtp.c b/main/rtp.c index 4f8846802..9d46f9720 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -2657,7 +2657,7 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int type_of_service, int class_of_servic return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc); } -void ast_rtp_update_source(struct ast_rtp *rtp) +void ast_rtp_new_source(struct ast_rtp *rtp) { if (rtp) { rtp->set_marker_bit = 1; |