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

channels/chan_sip.c

@@ -13097,7 +13097,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
 	/* Opus mandates 2 channels in rtpmap */
 	if (ast_format_cmp(format, ast_format_opus) == AST_FORMAT_CMP_EQUAL) {
 		ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u/2\r\n", rtp_code, mime, rate);
-	} else if ((35 <= rtp_code) || !(sip_cfg.compactheaders)) {
+	} else if ((AST_RTP_PT_LAST_STATIC < rtp_code) || !(sip_cfg.compactheaders)) {
 		ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, mime, rate);
 	}
 

include/asterisk/rtp_engine.h

@@ -81,6 +81,12 @@ extern "C" {
 /*! Maximum number of payload types RTP can support. */
 #define AST_RTP_MAX_PT 128
 
+/*!
+ * Last RTP payload type statically assigned, see
+ * http://www.iana.org/assignments/rtp-parameters
+ */
+#define AST_RTP_PT_LAST_STATIC 34
+
 /*! First dynamic RTP payload type */
 #define AST_RTP_PT_FIRST_DYNAMIC 96
 

main/rtp_engine.c

@@ -1348,28 +1348,31 @@ static int find_unused_payload(const struct ast_rtp_codecs *codecs)
 	 * https://tools.ietf.org/html/draft-roach-mmusic-unified-plan#section-3.2.1.2
 	 * https://tools.ietf.org/html/draft-wu-avtcore-dynamic-pt-usage#section-3
 	 */
-	res = find_unused_payload_in_range(codecs, MAX(ast_option_rtpptdynamic, 35),
+	res = find_unused_payload_in_range(
+		codecs, MAX(ast_option_rtpptdynamic, AST_RTP_PT_LAST_STATIC + 1),
 		AST_RTP_PT_LAST_REASSIGN, static_RTP_PT);
 	if (res != -1) {
 		return res;
 	}
 
-	/* Yet, reusing mappings below 35 is not supported in Asterisk because
-	 * when Compact Headers are activated, no rtpmap is send for those below
-	 * 35. If you want to use 35 and below
+	/* Yet, reusing mappings below AST_RTP_PT_LAST_STATIC (35) is not supported
+	 * in Asterisk because when Compact Headers are activated, no rtpmap is
+	 * send for those below 35. If you want to use 35 and below
 	 * A) do not use Compact Headers,
 	 * B) remove that code in chan_sip/res_pjsip, or
 	 * C) add a flag that this RTP Payload Type got reassigned dynamically
 	 *    and requires a rtpmap even with Compact Headers enabled.
 	 */
 	res = find_unused_payload_in_range(
-		codecs, MAX(ast_option_rtpptdynamic, 20), 35, static_RTP_PT);
+		codecs, MAX(ast_option_rtpptdynamic, 20),
+		AST_RTP_PT_LAST_STATIC + 1, static_RTP_PT);
 	if (res != -1) {
 		return res;
 	}
 
 	return find_unused_payload_in_range(
-		codecs, MAX(ast_option_rtpptdynamic, 0), 20, static_RTP_PT);
+		codecs, MAX(ast_option_rtpptdynamic, 0),
+		20, static_RTP_PT);
 }
 
 /*!

res/res_pjsip_sdp_rtp.c

@@ -447,6 +447,7 @@ static int set_caps(struct ast_sip_session *session,
 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 @@ static pjmedia_sdp_attr* generate_rtpmap_attr(struct ast_sip_session *session, p
 
 	snprintf(tmp, sizeof(tmp), "%d", rtp_code);
 	pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
+
+	if (rtp_code <= AST_RTP_PT_LAST_STATIC && 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 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 			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;
@@ -1287,12 +1291,10 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 				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);
 				attr = pjmedia_sdp_attr_create(pool, "fmtp", pj_cstr(&stmp, tmp));