allow users of RTP to use G726-32 AAL2 packing even when RFC3551 packing has been requested (Sipura/Grandstream ATAs and others will need this)

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37501 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 18 insertions, 18 deletions

diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index c5957e399..cefecc783 100644
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -882,7 +882,7 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
 		while (codec) {
 			ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
 			ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio",
-									iks_find_attrib(codec, "name"));
+						iks_find_attrib(codec, "name"), 0);
 			codec = iks_next(codec);
 		}
 		
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index df7db1b8a..5ad6cb733 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1882,7 +1882,7 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
 		if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
 			continue;
 		/* Note: should really look at the 'freq' and '#chans' params too */
-		ast_rtp_set_rtpmap_type(sub->rtp, codec, "audio", mimeSubtype);
+		ast_rtp_set_rtpmap_type(sub->rtp, codec, "audio", mimeSubtype, 0);
 	}
 
 	/* Now gather all of the codecs that were asked for: */
@@ -2081,7 +2081,7 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
 			if (codec > -1) {
 				snprintf(costr, sizeof(costr), " %d", codec);
 				strncat(m, costr, sizeof(m) - strlen(m) - 1);
-				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x));
+				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x, 0));
 				strncat(a, costr, sizeof(a) - strlen(a) - 1);
 			}
 		}
@@ -2095,7 +2095,7 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
 			if (codec > -1) {
 				snprintf(costr, sizeof(costr), " %d", codec);
 				strncat(m, costr, sizeof(m) - strlen(m) - 1);
-				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x));
+				snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x, 0));
 				strncat(a, costr, sizeof(a) - strlen(a) - 1);
 				if (x == AST_RTP_DTMF) {
 					/* Indicate we support DTMF...  Not sure about 16,
@@ -2140,7 +2140,7 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	snprintf(local, sizeof(local), "p:20");
 	for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
 		if (p->capability & x) {
-			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x));
+			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
 			strncat(local, tmp, sizeof(local) - strlen(local) - 1);
 		}
 	}
@@ -2170,7 +2170,7 @@ static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	snprintf(local, sizeof(local), "p:20");
 	for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
 		if (p->capability & x) {
-			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x));
+			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
 			strncat(local, tmp, sizeof(local) - strlen(local) - 1);
 		}
 	}
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 91aaef100..35ac561a6 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4719,9 +4719,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 			ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
 
 		/* Note: should really look at the 'freq' and '#chans' params too */
-		ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype);
+		ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype, 0);
 		if (p->vrtp)
-			ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype);
+			ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0);
 	}
 	
 	if (udptlportno != -1) {
@@ -4855,15 +4855,15 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 		char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ];
 
 		ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
-			ast_getformatname_multiple(s1, BUFSIZ, p->capability),
-			ast_getformatname_multiple(s2, BUFSIZ, newpeercapability),
-			ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
-			ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
+			    ast_getformatname_multiple(s1, BUFSIZ, p->capability),
+			    ast_getformatname_multiple(s2, BUFSIZ, newpeercapability),
+			    ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
+			    ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
 
 		ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
-			ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0),
-			ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0),
-			ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0));
+			    ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0, 0),
+			    ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
+			    ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
 	}
 	if (!newjointcapability) {
 		ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
@@ -5571,7 +5571,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
 
 	ast_build_string(m_buf, m_size, " %d", rtp_code);
 	ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
-			 ast_rtp_lookup_mime_subtype(1, codec),
+			 ast_rtp_lookup_mime_subtype(1, codec, 0),
 			 sample_rate);
 	if (codec == AST_FORMAT_G729A) {
 		/* Indicate that we don't support VAD (G.729 annex B) */
@@ -5727,13 +5727,13 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
 	int rtp_code;
 
 	if (debug)
-		ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format));
+		ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format, 0));
 	if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1)
 		return;
 
 	ast_build_string(m_buf, m_size, " %d", rtp_code);
 	ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
-			 ast_rtp_lookup_mime_subtype(0, format),
+			 ast_rtp_lookup_mime_subtype(0, format, 0),
 			 sample_rate);
 	if (format == AST_RTP_DTMF)
 		/* Indicate we support DTMF and FLASH... */