Merge "pjsip: Fix a few media bugs with reinvites and asymmetric payloads."

8 files changed, 74 insertions, 7 deletions

diff --git a/CHANGES b/CHANGES
index 5948360da..c9126e16b 100644
--- a/CHANGES
+++ b/CHANGES
@@ -57,6 +57,13 @@ res_pjsip
    res_pjsip_multihomed module has also been moved into core res_pjsip to ensure
    that messages are updated with the correct address information in all cases.
 
+chan_pjsip
+------------------
+ * The default behavior for RTP codecs has been changed. The sending codec will
+   now match the receiving codec. This can be turned off and behavior reverted
+   to asymmetric using the "asymmetric_rtp_codec" endpoint option. If this
+   option is set then the sending and received codec are allowed to differ.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ----------
 ------------------------------------------------------------------------------
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index ea06d6757..b0dba1b47 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -219,9 +219,7 @@ static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *cha
 /*! \brief Function called by RTP engine to get peer capabilities */
 static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
 {
-	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
-
-	ast_format_cap_append_from_cap(result, channel->session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
+	ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
 }
 
 /*! \brief Destructor function for \ref transport_info_data */
@@ -725,15 +723,28 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
 
 	session = channel->session;
 
-	if (ast_format_cap_iscompatible_format(session->endpoint->media.codecs, f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
-		ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when endpoint '%s' is not configured for it\n",
-			ast_format_get_name(f->subclass.format), ast_channel_name(ast),
-			ast_sorcery_object_get_id(session->endpoint));
+	if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
+		ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
+			ast_format_get_name(f->subclass.format), ast_channel_name(ast));
 
 		ast_frfree(f);
 		return &ast_null_frame;
 	}
 
+	if (!session->endpoint->asymmetric_rtp_codec &&
+		ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
+		/* For maximum compatibility we ensure that the write format matches that of the received media */
+		ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
+			ast_format_get_name(f->subclass.format), ast_channel_name(ast),
+			ast_format_get_name(ast_channel_rawwriteformat(ast)));
+		ast_channel_set_rawwriteformat(ast, f->subclass.format);
+		ast_set_write_format(ast, ast_channel_writeformat(ast));
+
+		if (ast_channel_is_bridged(ast)) {
+			ast_channel_set_unbridged_nolock(ast, 1);
+		}
+	}
+
 	if (session->dsp) {
 		int dsp_features;
 
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 3bb9dc5bb..6595423c9 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -759,6 +759,8 @@
                                 ; rather than advertising all joint codec capabilities. This
                                 ; limits the other side's codec choice to exactly what we prefer.
                                 ; default is no.
+;asymmetric_rtp_codec= ; Allow the sending and receiving codec to differ and
+                       ; not be automatically matched (default: "no")
 
 ;==========================AUTH SECTION OPTIONS=========================
 ;[auth]
diff --git a/contrib/ast-db-manage/config/versions/4468b4a91372_add_pjsip_asymmetric_rtp_codec.py b/contrib/ast-db-manage/config/versions/4468b4a91372_add_pjsip_asymmetric_rtp_codec.py
new file mode 100644
index 000000000..c121495e2
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/4468b4a91372_add_pjsip_asymmetric_rtp_codec.py
@@ -0,0 +1,31 @@
+"""add pjsip asymmetric rtp codec
+
+Revision ID: 4468b4a91372
+Revises: a6ef36f1309
+Create Date: 2016-10-25 10:57:20.808815
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '4468b4a91372'
+down_revision = 'a6ef36f1309'
+
+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('asymmetric_rtp_codec', yesno_values))
+
+
+def downgrade():
+    op.drop_column('ps_endpoints', 'asymmetric_rtp_codec')
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 92bdabb66..894ea76f5 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -759,6 +759,8 @@ struct ast_sip_endpoint {
 	char *contact_user;
 	/*! Whether to response SDP offer with single most preferred codec. */
 	unsigned int preferred_codec_only;
+	/*! Do we allow an asymmetric RTP codec? */
+	unsigned int asymmetric_rtp_codec;
 };
 
 /*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 39c365aa1..916c464a1 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -925,6 +925,14 @@
 						On outbound requests, force the user portion of the Contact header to this value.
 					</para></description>
 				</configOption>
+                                <configOption name="asymmetric_rtp_codec" default="no">
+                                        <synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
+                                        <description><para>
+                                                When set to "yes" the codec in use for sending will be allowed to differ from
+                                                that of the received one. PJSIP will not automatically switch the sending one
+                                                to the receiving one.
+                                        </para></description>
+                                </configOption>
 			</configObject>
 			<configObject name="auth">
 				<synopsis>Authentication type</synopsis>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 186646759..00c223330 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1938,6 +1938,7 @@ int ast_res_pjsip_initialize_configuration(void)
 	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));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "asymmetric_rtp_codec", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, asymmetric_rtp_codec));
 
 	if (ast_sip_initialize_sorcery_transport()) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index a69aa1a74..13a71d478 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -385,6 +385,11 @@ static int set_caps(struct ast_sip_session *session,
 				session->dsp = NULL;
 			}
 		}
+
+		if (ast_channel_is_bridged(session->channel)) {
+			ast_channel_set_unbridged_nolock(session->channel, 1);
+		}
+
 		ast_channel_unlock(session->channel);
 	}