res/res_pjsip: Add preferred_codec_only config to pjsip endpoint.

This patch add config to pjsip by endpoint. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. ASTERISK-26317 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
2025-09-05 12:16:00 +00:00 · 2016-08-30 11:26:03 +08:00
parent d3c4b901d4
commit 2a50c29101
8 changed files with 63 additions and 4 deletions
--- a/7
+++ b/7
@@ -13,6 +13,13 @@
 ------------------------------------------------------------------------------


+res_pjsip
+------------------
+ * Added endpoint configuration parameter "preferred_codec_only".
+   This allow asterisk response to a SIP invite with the single most
+   preferred codec rather than advertising all joint codec capabilities.
+   This limits the other side's codec choice to exactly what we prefer.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ----------
 ------------------------------------------------------------------------------
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -764,6 +764,10 @@
                   ; "0" or not enabled)
 ;contact_user= ; On outgoing requests, force the user portion of the Contact
               ; header to this value (default: "")
+;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
+                                ; rather than advertising all joint codec capabilities. This
+                                ; limits the other side's codec choice to exactly what we prefer.
+                                ; default is no.

 ;==========================AUTH SECTION OPTIONS=========================
 ;[auth]
--- a/contrib/ast-db-manage/config/versions/7f3e21abe318_add_preferred_codec_only_option_to_pjsip.py
+++ b/contrib/ast-db-manage/config/versions/7f3e21abe318_add_preferred_codec_only_option_to_pjsip.py
@@ -0,0 +1,30 @@
+"""add preferred_codec_only option to pjsip
+
+Revision ID: 7f3e21abe318
+Revises: 4e2493ef32e6
+Create Date: 2016-09-02 11:00:23.534748
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '7f3e21abe318'
+down_revision = '4e2493ef32e6'
+
+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('preferred_codec_only', yesno_values))
+
+def downgrade():
+    op.drop_column('ps_endpoints', 'preferred_codec_only')
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -757,6 +757,8 @@ struct ast_sip_endpoint {
 	unsigned int faxdetect_timeout;
 	/*! Override the user on the outgoing Contact header with this value. */
 	char *contact_user;
+	/*! Whether to response SDP offer with single most preferred codec. */
+	unsigned int preferred_codec_only;
 };

 /*!
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -833,6 +833,9 @@
 						have this accountcode set on it.
 					</para></description>
 				</configOption>
+				<configOption name="preferred_codec_only" default="no">
+					<synopsis>Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer.</synopsis>
+				</configOption>
 				<configOption name="rtp_keepalive">
 					<synopsis>Number of seconds between RTP comfort noise keepalive packets.</synopsis>
 					<description><para>
@@ -2022,6 +2025,9 @@
 				<parameter name="Accountcode">
 					<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='accountcode']/synopsis/node())"/></para>
 				</parameter>
+				<parameter name="PreferredCodecOnly">
+					<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='preferred_codec_only']/synopsis/node())"/></para>
+				</parameter>
 				<parameter name="DeviceState">
 					<para>The aggregate device state for this endpoint.</para>
 				</parameter>
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1933,6 +1933,7 @@ int ast_res_pjsip_initialize_configuration(void)
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_acl", "", endpoint_acl_handler, contact_acl_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "subscribe_context", "", OPT_CHAR_ARRAY_T, 0, CHARFLDSET(struct ast_sip_endpoint, subscription.context));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_user", "", contact_user_handler, contact_user_to_str, NULL, 0, 0);
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "preferred_codec_only", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, preferred_codec_only));

 	if (ast_sip_initialize_sorcery_transport()) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -360,8 +360,13 @@ static int set_caps(struct ast_sip_session *session,
 		ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
 			AST_MEDIA_TYPE_UNKNOWN);
 		ast_format_cap_remove_by_type(caps, media_type);
+		if (session->endpoint->preferred_codec_only){
+			struct ast_format *preferred_fmt = ast_format_cap_get_format(joint, 0);
+			ast_format_cap_append(caps, preferred_fmt, 0);
+			ao2_ref(preferred_fmt, -1);
+		} else {
 			ast_format_cap_append_from_cap(caps, joint, media_type);
-
+		}
 		/*
 		 * Apply the new formats to the channel, potentially changing
 		 * raw read/write formats and translation path while doing so.
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1247,7 +1247,9 @@ int ast_sip_session_create_invite(struct ast_sip_session *session, pjsip_tx_data
 	pjsip_inv_set_local_sdp(session->inv_session, offer);
 	pjmedia_sdp_neg_set_prefer_remote_codec_order(session->inv_session->neg, PJ_FALSE);
 #ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
+	if (!session->endpoint->preferred_codec_only) {
 		pjmedia_sdp_neg_set_answer_multiple_codecs(session->inv_session->neg, PJ_TRUE);
+	}
 #endif

 	/*
@@ -2141,7 +2143,9 @@ static int new_invite(void *data)
 		pjsip_inv_set_local_sdp(invite->session->inv_session, local);
 		pjmedia_sdp_neg_set_prefer_remote_codec_order(invite->session->inv_session->neg, PJ_FALSE);
 #ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
+		if (!invite->session->endpoint->preferred_codec_only) {
 			pjmedia_sdp_neg_set_answer_multiple_codecs(invite->session->inv_session->neg, PJ_TRUE);
+		}
 #endif
 	}