[Asterisk-code-review] res pjsip sdp rtp: No rtpmap for static RTP payload IDs in SDP. (asterisk[master])

[Alexander Traud]

Alexander Traud has uploaded a new change for review. ( https://gerrit.asterisk.org/5431 )

Change subject: res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP.
......................................................................

res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP.

This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.

ASTERISK-26932 #close

Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
---
M res/res_pjsip_sdp_rtp.c
1 file changed, 10 insertions(+), 8 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/31/5431/1

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 701edc3..022c9e3 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -447,6 +447,7 @@
 static pjmedia_sdp_attr* generate_rtpmap_attr(struct ast_sip_session *session, pjmedia_sdp_media *media, pj_pool_t *pool,
 					      int rtp_code, int asterisk_format, struct ast_format *format, int code)
 {
+	extern pj_bool_t pjsip_use_compact_form;
 	pjmedia_sdp_rtpmap rtpmap;
 	pjmedia_sdp_attr *attr = NULL;
 	char tmp[64];
@@ -455,6 +456,11 @@
 
 	snprintf(tmp, sizeof(tmp), "%d", rtp_code);
 	pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
+
+	if (35 > rtp_code && pjsip_use_compact_form) {
+		return NULL;
+	}
+
 	rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1];
 	rtpmap.clock_rate = ast_rtp_lookup_sample_rate2(asterisk_format, format, code);
 	pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options));
@@ -1254,11 +1260,9 @@
 			continue;
 		}
 
-		if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
-			ao2_ref(format, -1);
-			continue;
+		if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
+			media->attr[media->attr_count++] = attr;
 		}
-		media->attr[media->attr_count++] = attr;
 
 		if ((attr = generate_fmtp_attr(pool, format, rtp_code))) {
 			media->attr[media->attr_count++] = attr;
@@ -1287,11 +1291,9 @@
 				continue;
 			}
 
-			if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 0, NULL, index))) {
-				continue;
+			if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 0, NULL, index))) {
+				media->attr[media->attr_count++] = attr;
 			}
-
-			media->attr[media->attr_count++] = attr;
 
 			if (index == AST_RTP_DTMF) {
 				snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code);

-- 
To view, visit https://gerrit.asterisk.org/5431
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Gerrit-MessageType: newchange
Gerrit-Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>