Merge "pjsip: Fix a few media bugs with reinvites and asymmetric payloads." into 14

changes/16/4216/1
Joshua Colp 9 years ago committed by Gerrit Code Review
commit 2a6d5fb271

@ -21,6 +21,13 @@ 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 ----------
------------------------------------------------------------------------------

@ -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;

@ -755,6 +755,8 @@
; "0" or not enabled)
;contact_user= ; On outgoing requests, force the user portion of the Contact
; header to this value (default: "")
;asymmetric_rtp_codec= ; Allow the sending and receiving codec to differ and
; not be automatically matched (default: "no")
;==========================AUTH SECTION OPTIONS=========================
;[auth]

@ -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')

@ -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;
/*! Do we allow an asymmetric RTP codec? */
unsigned int asymmetric_rtp_codec;
};
/*!

@ -922,6 +922,14 @@
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>

@ -1937,6 +1937,7 @@ 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", "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");

@ -380,6 +380,11 @@ 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);
}

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