[Asterisk-code-review] res pjsip sdp rtp: Only enable abs-send-time when WebRTC is ... (asterisk[master])

Joshua C. Colp asteriskteam at digium.com
Mon Jan 14 08:03:28 CST 2019


Joshua C. Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/10858 )

Change subject: res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.
......................................................................

res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.

For video streams it was possible for the abs-send-time information
to be placed into RTP streams even if not negotiated. Depending on
the endpoint in use this could cause video to not flow.

We now only enable abs-send-time for negotiation if WebRTC is enabled.

ASTERISK-28230

Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c
---
M res/res_pjsip_sdp_rtp.c
1 file changed, 3 insertions(+), 1 deletion(-)

Approvals:
  Benjamin Keith Ford: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Joshua C. Colp: Approved for Submit



diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 2889615..c18627e 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -272,7 +272,9 @@
 		ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_RECV, session->endpoint->media.webrtc);
 		ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_SEND, session->endpoint->media.webrtc);
 		ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_REMB, session->endpoint->media.webrtc);
-		enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
+		if (session->endpoint->media.webrtc) {
+			enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
+		}
 		if (session->endpoint->media.tos_video || session->endpoint->media.cos_video) {
 			ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
 					session->endpoint->media.cos_video, "SIP RTP Video");

-- 
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c
Gerrit-Change-Number: 10858
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua C. Colp <jcolp at digium.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Friendly Automation (1000185)
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua C. Colp <jcolp at digium.com>
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