Merge "res_pjsip: Add option to force G.726 to be treated as AAL2 packed."

CHANGES

@@ -186,6 +186,12 @@ AMI
  * A new ContactStatus event has been added that reflects res_pjsip contact
    lifecycle changes:  Created, Removed, Reachable, Unreachable, Unknown.
 
+res_pjsip
+------------------
+* A new 'g726_non_standard' endpoint option has been added that, when set to
+  'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
+  is AAL2 packed on the channel.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
 ------------------------------------------------------------------------------

configs/samples/pjsip.conf.sample

@@ -658,6 +658,11 @@
                         ; "no")
 ;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call
 								; if not possible.
+;g726_non_standard=no   ; When set to "yes" and an endpoint negotiates g.726
+                        ; audio then g.726 for AAL2 packing order is used contrary
+                        ; to what is recommended in RFC3551. Note, 'g726aal2' also
+                        ; needs to be specified in the codec allow list
+                        ; (default: "no")
 ;inband_progress=no     ; Determines whether chan_pjsip will indicate ringing
                         ; using inband progress (default: "no")
 ;call_group=    ; The numeric pickup groups for a channel (default: "")

contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.py

@@ -0,0 +1,30 @@
+"""add g726_non_standard
+
+Revision ID: 28b8e71e541f
+Revises: a541e0b5e89
+Create Date: 2015-06-12 16:07:08.609628
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '28b8e71e541f'
+down_revision = 'a541e0b5e89'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+    ############################# Enums ##############################
+
+    # yesno_values have already been created, so use postgres enum object
+    # type to get around "already created" issue - works okay with mysql
+    yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+    op.add_column('ps_endpoints', sa.Column('g726_non_standard', yesno_values))
+
+
+def downgrade():
+    op.drop_column('ps_endpoints', 'g726_non_standard')

include/asterisk/res_pjsip.h

@@ -557,6 +557,8 @@ struct ast_sip_endpoint_media_configuration {
 	unsigned int tos_video;
 	/*! Priority for video streams */
 	unsigned int cos_video;
+	/*! Is g.726 packed in a non standard way */
+	unsigned int g726_non_standard;
 };
 
 /*!

res/res_pjsip.c

@@ -471,6 +471,15 @@
 						set to <literal>sdes</literal> or <literal>dtls</literal>.
 					</para></description>
 				</configOption>
+				<configOption name="g726_non_standard" default="no">
+					<synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
+					<description><para>
+                                                When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
+                                                packing order instead of what is recommended by RFC3551. Since this essentially
+                                                replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
+                                                specified in the endpoint's allowed codec list.
+					</para></description>
+				</configOption>
 				<configOption name="inband_progress" default="no">
 					<synopsis>Determines whether chan_pjsip will indicate ringing using inband
 					    progress.</synopsis>

res/res_pjsip/pjsip_configuration.c

@@ -1923,6 +1923,7 @@ int ast_res_pjsip_initialize_configuration(void)
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "dtls_fingerprint", "", dtls_handler, dtlsfingerprint_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "srtp_tag_32", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.srtp_tag_32));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "media_encryption_optimistic", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.encryption_optimistic));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "g726_non_standard", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.g726_non_standard));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "redirect_method", "user", redirect_handler, NULL, NULL, 0, 0);
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "set_var", "", set_var_handler, set_var_to_str, set_var_to_vl, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "message_context", "", OPT_STRINGFIELD_T, 1, STRFLDSET(struct ast_sip_endpoint, message_context));

res/res_pjsip_sdp_rtp.c

@@ -155,6 +155,8 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
 	char name[256];
 	char media[20];
 	char fmt_param[256];
+	enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
+		AST_RTP_OPT_G726_NONSTANDARD : 0;
 
 	ast_rtp_codecs_payloads_initialize(codecs);
 
@@ -176,9 +178,10 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
                 if (strcmp(name,"telephone-event") == 0) {
                         tel_event++;
                 }
+
 		ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
 		ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]),
-							     media, name, 0, rtpmap->clock_rate);
+							     media, name, options, rtpmap->clock_rate);
 		/* Look for an optional associated fmtp attribute */
 		if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) {
 			continue;
@@ -304,18 +307,20 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
 	return 0;
 }
 
-static pjmedia_sdp_attr* generate_rtpmap_attr(pjmedia_sdp_media *media, pj_pool_t *pool, int rtp_code,
-					      int asterisk_format, struct ast_format *format, int code)
+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)
 {
 	pjmedia_sdp_rtpmap rtpmap;
 	pjmedia_sdp_attr *attr = NULL;
 	char tmp[64];
+	enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
+		AST_RTP_OPT_G726_NONSTANDARD : 0;
 
 	snprintf(tmp, sizeof(tmp), "%d", rtp_code);
 	pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
 	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, 0));
+	pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options));
 	rtpmap.param.slen = 0;
 	rtpmap.param.ptr = NULL;
 
@@ -1051,7 +1056,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 			continue;
 		}
 
-		if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 1, format, 0))) {
+		if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
 			ao2_ref(format, -1);
 			continue;
 		}
@@ -1076,7 +1081,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 				continue;
 			}
 
-			if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 0, NULL, index))) {
+			if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 0, NULL, index))) {
 				continue;
 			}