pjsip: fix support for allow=all

This change adds improvements to support for allow=all in
pjsip.conf so that it functions as intended.  Previously,
the allow/disallow socery configuration would set & clear
codecs from the media.codecs and media.prefs list, but if
all was specified the prefs list was not updated.  Then a
call would fail when create_outgoing_sdp_stream() created
an SDP with no audio codecs.

A new function ast_codec_pref_append_all() is provided to
add all codecs to the prefs list - only those not already
on the list.  This enables the configuration to specify a
codec preference, but still add all codecs, and even then
remove some codecs, as shown in this example:

allow = ulaw, alaw, all, !g729, !g723

Also, the display order of allow in cli output is updated
to match the configuration by using prefs instead of caps
when generating a human readable string.

Finally, a change to create_outgoing_sdp_stream() skips a
codec when it does not have a payload code instead of the
call failing.

(closes issue ASTERISK-23018)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3131/



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405875 65c4cc65-6c06-0410-ace0-fbb531ad65f3

include/asterisk/format_pref.h

@@ -69,6 +69,9 @@ struct ast_format *ast_codec_pref_index(struct ast_codec_pref *pref, int index,
 /*! \brief Remove audio a codec from a preference list */
 void ast_codec_pref_remove(struct ast_codec_pref *pref, struct ast_format *format);
 
+/*! \brief Append all codecs to a preference list, without disturbing existing order */
+void ast_codec_pref_append_all(struct ast_codec_pref *pref);
+
 /*! \brief Append a audio codec to a preference list, removing it first if it was already there
 */
 int ast_codec_pref_append(struct ast_codec_pref *pref, struct ast_format *format);

main/format_pref.c

@@ -143,6 +143,42 @@ void ast_codec_pref_remove(struct ast_codec_pref *pref, struct ast_format *forma
 	ast_format_list_destroy(f_list);
 }
 
+/*! \brief Append all codecs to a preference list, without distrubing existing order */
+void ast_codec_pref_append_all(struct ast_codec_pref *pref)
+{
+	int x, y, found;
+	size_t f_len = 0;
+	const struct ast_format_list *f_list = ast_format_list_get(&f_len);
+
+	/* leave any existing entries, and don't create duplicates (e.g. allow=ulaw,all) */
+	for (x = 0; x < f_len; x++) {
+		/* x = codec to add */
+		found = 0;
+		for (y = 0; y < f_len; y++) {
+			/* y = scan of existing preferences */
+			if (!pref->order[y]) {
+				break;
+			}
+			if (x + 1 == pref->order[y]) {
+				found = 1;
+				break;
+			}
+		}
+		if (found) {
+			continue;
+		}
+		for (; y < f_len; y++) {
+			/* add x to the end of y */
+			if (!pref->order[y])
+			{
+				pref->order[y] = x + 1;
+				ast_format_copy(&pref->formats[y], &f_list[x].format);
+				break;
+			}
+		}
+	}
+}
+
 /*! \brief Append codec to list */
 int ast_codec_pref_append(struct ast_codec_pref *pref, struct ast_format *format)
 {

main/frame.c

@@ -866,6 +866,8 @@ int ast_parse_allow_disallow(struct ast_codec_pref *pref, struct ast_format_cap
 				}
 			} else if (!iter_allowing) {
 				memset(pref, 0, sizeof(*pref));
+			} else {
+				ast_codec_pref_append_all(pref);
 			}
 		}
 	}

main/sorcery.c

@@ -43,6 +43,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/taskprocessor.h"
 #include "asterisk/threadpool.h"
 #include "asterisk/json.h"
+#include "asterisk/format_pref.h"
 
 /* To prevent DEBUG_FD_LEAKS from interfering with things we undef open and close */
 #undef open
@@ -222,8 +223,9 @@ static int chararray_handler_fn(const void *obj, const intptr_t *args, char **bu
 static int codec_handler_fn(const void *obj, const intptr_t *args, char **buf)
 {
 	char tmp_buf[256];
-	struct ast_format_cap **cap = (struct ast_format_cap **)(obj + args[1]);
-	return !(*buf = ast_strdup(ast_getformatname_multiple(tmp_buf, sizeof(tmp_buf), *cap)));
+	struct ast_codec_pref *pref = (struct ast_codec_pref *)(obj + args[0]);
+	ast_codec_pref_string(pref, tmp_buf, sizeof(tmp_buf));
+	return !(*buf = ast_strdup(tmp_buf));
 }
 
 static sorcery_field_handler sorcery_field_default_handler(enum aco_option_type type)

res/res_pjsip_sdp_rtp.c

@@ -936,7 +936,8 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 		}
 
 		if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, &format, 0)) == -1) {
-			return -1;
+			ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n",ast_getformatname(&format));
+			continue;
 		}
 
 		if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 1, &format, 0))) {