diff --git a/CHANGES b/CHANGES
index 5857165c8d75e3a66d967b0243a95ae8c35af4b4..6032dd8a053a6471f9e436ce819a6687b4be5ab2 100644
--- a/CHANGES
+++ b/CHANGES
@@ -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 ----------
------------------------------------------------------------------------------
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 0d1c03909f1cf5154c97929f48e085a5d1581e99..e6b32495ba36f5864c41de7d143f24acec8f4725 100644
--- 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]
diff --git 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
new file mode 100644
index 0000000000000000000000000000000000000000..083d08966e838c5dde0d2dc18a83b56df11e155a
--- /dev/null
+++ 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')
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 4cede43919b3b9d770237393d4122f8c6a57792b..8a5ad29c59810d9f745320c0199d394ffdce59b8 100644
--- 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;
};
/*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 34edc8ca511ccd7e1e8c6fb12accadce4395afc0..7bb10c07f01b59035ebceca2653d6283f1f2181a 100644
--- 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>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 333be71438a735b00fb0772a45f5b1ce71d7cb14..97c357a9e1ce61716c5fe65777dab618f72e5297 100644
--- 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");
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 6610ef126c365f3358c2c86a1472920e226780ff..68d5fdb56389211bcd1cc4d6b8e8a37087a4d882 100644
--- 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);
- ast_format_cap_append_from_cap(caps, joint, 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.
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 315393fdb895e6005eddedfd1c99cce72c2da0e2..0ab45c74e1306e5a8f8f9a5797639e28efbb7b96 100644
--- 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
- pjmedia_sdp_neg_set_answer_multiple_codecs(session->inv_session->neg, PJ_TRUE);
+ 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
- pjmedia_sdp_neg_set_answer_multiple_codecs(invite->session->inv_session->neg, PJ_TRUE);
+ if (!invite->session->endpoint->preferred_codec_only) {
+ pjmedia_sdp_neg_set_answer_multiple_codecs(invite->session->inv_session->neg, PJ_TRUE);
+ }
#endif
}