diff --git a/CHANGES b/CHANGES
index 622973c48f0090448ca2c9119d627b9cb6cf4869..b1e398576237693c6381159b40ad8e5f40ea632d 100644
--- a/CHANGES
+++ b/CHANGES
@@ -47,6 +47,13 @@ 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 00d4a1452fd34f3bc120ea539ae61721f74baf54..0a4e5c266e5fb619f660b09dae1bee951d06600c 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 */
@@ -704,15 +702,28 @@ 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 3bb9dc5bb72586eb31609a06fde4face3549575d..6595423c990b8723624f4a3d470e583fc44d3449 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 0000000000000000000000000000000000000000..c121495e2d2d57664ffd464e0a63ceeed5940919
--- /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 92bdabb662f772ee4b71482600fba9de3a664cef..894ea76f563166e7a7bd0c213ff10eba60d542c1 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 39c365aa1acbbdec6b951213f014977f8dd30d0a..916c464a11708c0ad5838a9d2b9a3aaa44817999 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 18664675999a8c320c38e3211210f79814c58ff3..00c22333078249b9e452b4d91fe6320a82430fe6 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 a69aa1a74383a8ae713cb3b53c13a9c1884a151e..13a71d4785c557a8f3460aebb9b9aff3d0cc7365 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);
}