[asterisk-commits] res rtp asterisk: Make P2P bridge Asymmetric codec aware (asterisk[15])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Aug 9 15:39:06 CDT 2017
Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/6187 )
Change subject: res_rtp_asterisk: Make P2P bridge Asymmetric codec aware
......................................................................
res_rtp_asterisk: Make P2P bridge Asymmetric codec aware
Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not. If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed
ASTERISK-26745 #close
Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
---
M channels/chan_pjsip.c
M include/asterisk/rtp_engine.h
M res/res_pjsip_sdp_rtp.c
M res/res_rtp_asterisk.c
4 files changed, 37 insertions(+), 7 deletions(-)
Approvals:
Joshua Colp: Looks good to me, but someone else must approve; Approved for Submit
George Joseph: Looks good to me, approved
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index ebda6c7..4a24fa6 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -797,14 +797,13 @@
return f;
}
- 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));
+ session = channel->session;
- ast_frfree(f);
- return &ast_null_frame;
- }
-
+ /*
+ * Asymmetric RTP only has one native format set at a time.
+ * Therefore we need to update the native format to the current
+ * raw read format BEFORE the native format check
+ */
if (!session->endpoint->asymmetric_rtp_codec &&
ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
struct ast_format_cap *caps;
@@ -831,6 +830,14 @@
}
}
+ 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->dsp) {
int dsp_features;
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index d030bdb..3ceac84 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -120,6 +120,8 @@
AST_RTP_PROPERTY_STUN,
/*! Enable RTCP support */
AST_RTP_PROPERTY_RTCP,
+ /*! Enable Asymmetric RTP Codecs */
+ AST_RTP_PROPERTY_ASYMMETRIC_CODEC,
/*!
* \brief Maximum number of RTP properties supported
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 77cd807..b082b1d 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -202,6 +202,7 @@
}
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->media.rtp.symmetric);
+ ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_ASYMMETRIC_CODEC, session->endpoint->asymmetric_rtp_codec);
if (!session->endpoint->media.rtp.ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
ice->stop(session_media->rtp);
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 65ab902..0ff2b55 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -313,6 +313,7 @@
void *data;
struct ast_rtcp *rtcp;
struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
+ unsigned int asymmetric_codec; /*!< Indicate if asymmetric send/receive codecs are allowed */
struct ast_rtp_instance *bundled; /*!< The RTP instance we are bundled to */
int stream_num; /*!< Stream num for this RTP instance */
@@ -5113,6 +5114,23 @@
return -1;
}
+
+ ao2_replace(rtp->lastrxformat, payload_type->format);
+ ao2_replace(bridged->lasttxformat, payload_type->format);
+
+ /*
+ * If bridged peer has already received rtp, perform the asymmetric codec check
+ * if that feature has been activated
+ */
+ if (!bridged->asymmetric_codec && bridged->lastrxformat != ast_format_none) {
+ if (ast_format_cmp(bridged->lasttxformat, bridged->lastrxformat) == AST_FORMAT_CMP_NOT_EQUAL) {
+ ast_debug(1, "Asymmetric RTP codecs detected (TX: %s, RX: %s) sending frame to core\n",
+ ast_format_get_name(bridged->lasttxformat),
+ ast_format_get_name(bridged->lastrxformat));
+ return -1;
+ }
+ }
+
/* If the marker bit has been explicitly set turn it on */
if (ast_test_flag(rtp, FLAG_NEED_MARKER_BIT)) {
mark = 1;
@@ -5797,6 +5815,8 @@
rtp->rtcp = NULL;
}
}
+ } else if (property == AST_RTP_PROPERTY_ASYMMETRIC_CODEC) {
+ rtp->asymmetric_codec = value;
}
}
--
To view, visit https://gerrit.asterisk.org/6187
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: 15
Gerrit-MessageType: merged
Gerrit-Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
Gerrit-Change-Number: 6187
Gerrit-PatchSet: 1
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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