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

diff --git a/CHANGES b/CHANGES
index 5857165c8d75e3a66d967b0243a95ae8c35af4b4..6032dd8a053a6471f9e436ce819a6687b4be5ab2 100644 (file)
--- 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 (file)
--- 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 (file)
index 0000000..083d089
--- /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 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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 (file)
--- 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
        }