[Asterisk-code-review] pjsip: Fix a few media bugs with reinvites and asymmetric pa... (asterisk[14])
Joshua Colp
asteriskteam at digium.com
Thu Oct 27 18:08:45 CDT 2016
Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/4173 )
Change subject: pjsip: Fix a few media bugs with reinvites and asymmetric payloads.
......................................................................
pjsip: Fix a few media bugs with reinvites and asymmetric payloads.
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
---
M CHANGES
M channels/chan_pjsip.c
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/4468b4a91372_add_pjsip_asymmetric_rtp_codec.py
M include/asterisk/res_pjsip.h
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
8 files changed, 74 insertions(+), 7 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
George Joseph: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
diff --git a/CHANGES b/CHANGES
index a20f2ad..1ce7422 100644
--- a/CHANGES
+++ b/CHANGES
@@ -21,6 +21,13 @@
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 00d4a14..0a4e5c2 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -219,9 +219,7 @@
/*! \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 @@
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 eda8022..9611ca5 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -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]
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 0000000..c121495
--- /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 9731fa6..7c7c3c7 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -757,6 +757,8 @@
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;
};
/*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 6b22c66..5d422d8 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -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>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index d8ae9e0..f7a4fdc 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1937,6 +1937,7 @@
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");
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 7937972..ad1d72f 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -380,6 +380,11 @@
session->dsp = NULL;
}
}
+
+ if (ast_channel_is_bridged(session->channel)) {
+ ast_channel_set_unbridged_nolock(session->channel, 1);
+ }
+
ast_channel_unlock(session->channel);
}
--
To view, visit https://gerrit.asterisk.org/4173
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
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