[Asterisk-code-review] res pjsip sdp rtp: Properly set packetization on the codec (asterisk[13])

[Robert Cripps]

Robert Cripps has uploaded this change for review. ( https://gerrit.asterisk.org/10495


Change subject: res_pjsip_sdp_rtp: Properly set packetization on the codec
......................................................................

res_pjsip_sdp_rtp: Properly set packetization on the codec

In the transcode case (Alaw to Ulaw) the ptime was being discarded.

res_rtp_asterisk:

In the native bridge case the framing was never checked.

ASTERISK-28110 #close

Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607
---
M res/res_pjsip_sdp_rtp.c
M res/res_rtp_asterisk.c
2 files changed, 11 insertions(+), 1 deletion(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/95/10495/1

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 7c7040e..d74c90c 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -392,7 +392,10 @@
 	if (!ast_format_cap_count(joint)) {
 		struct ast_str *usbuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
 		struct ast_str *thembuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
-
+		if(codecs.framing) {
+			/* Copy the codecs framing ms even though there is no match */
+			ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(session_media->rtp),codecs.framing);
+		}
 		ast_rtp_codecs_payloads_destroy(&codecs);
 		ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
 			session_media->stream_type,
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 7b006ce..26d4845 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -5160,6 +5160,13 @@
 		return -1;
 	}
 
+	/* In p2p mode the framing ms values for both must be equal. If bridged is zero we don't care */
+	unsigned framing1 = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance1));
+	unsigned framing = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
+	if (framing1 != 0 && framing != framing1) {
+		return -1;
+	}
+
 	/* Otherwise adjust bridged payload to match */
 	bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type->asterisk_format, payload_type->format, payload_type->rtp_code);
 

-- 
To view, visit https://gerrit.asterisk.org/10495
To unsubscribe, or for help writing mail filters, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: newchange
Gerrit-Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607
Gerrit-Change-Number: 10495
Gerrit-PatchSet: 1
Gerrit-Owner: Robert Cripps <rcripps at voxbone.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20181017/b868b7a3/attachment-0001.html>