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res_pjsip_sdp_rtp.c: Support keepalive for video streams. 

ASTERISK-28890 #close

Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1
This commit is contained in:
Luke Escude 2022-01-19 17:33:51 -05:00 committed by Joshua Colp
parent 8c230ebb87
commit 14156f9827
1 changed files with 3 additions and 2 deletions
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5
res/res_pjsip_sdp_rtp.c
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@ -2089,7 +2089,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
ast_rtp_instance_activate(session_media->rtp);
/* audio stream handles music on hold */
if (media_type != AST_MEDIA_TYPE_AUDIO) {
if (media_type != AST_MEDIA_TYPE_AUDIO && media_type != AST_MEDIA_TYPE_VIDEO) {
if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
@ -2121,7 +2121,8 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
session_media->encryption = session->endpoint->media.rtp.encryption;
if (session->endpoint->media.rtp.keepalive > 0 &&
session_media->type == AST_MEDIA_TYPE_AUDIO) {
(session_media->type == AST_MEDIA_TYPE_AUDIO ||
session_media->type == AST_MEDIA_TYPE_VIDEO)) {
ast_rtp_instance_set_keepalive(session_media->rtp, session->endpoint->media.rtp.keepalive);
/* Schedule the initial keepalive early in case this is being used to punch holes through
* a NAT. This way there won't be an awkward delay before media starts flowing in some