[Asterisk-code-review] res pjsip: Add option to force G.726 to be treated as AAL2 p... (asterisk[master])
Joshua Colp
asteriskteam at digium.com
Tue Jun 16 07:51:30 CDT 2015
Joshua Colp has submitted this change and it was merged.
Change subject: res_pjsip: Add option to force G.726 to be treated as AAL2 packed.
......................................................................
res_pjsip: Add option to force G.726 to be treated as AAL2 packed.
Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
---
M CHANGES
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.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
7 files changed, 64 insertions(+), 6 deletions(-)
Approvals:
Mark Michelson: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, approved; Verified
diff --git a/CHANGES b/CHANGES
index 281d059..d2fa84c 100644
--- a/CHANGES
+++ b/CHANGES
@@ -186,6 +186,12 @@
* A new ContactStatus event has been added that reflects res_pjsip contact
lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
+res_pjsip
+------------------
+* A new 'g726_non_standard' endpoint option has been added that, when set to
+ 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
+ is AAL2 packed on the channel.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
------------------------------------------------------------------------------
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 276e214..24ff327 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -658,6 +658,11 @@
; "no")
;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call
; if not possible.
+;g726_non_standard=no ; When set to "yes" and an endpoint negotiates g.726
+ ; audio then g.726 for AAL2 packing order is used contrary
+ ; to what is recommended in RFC3551. Note, 'g726aal2' also
+ ; needs to be specified in the codec allow list
+ ; (default: "no")
;inband_progress=no ; Determines whether chan_pjsip will indicate ringing
; using inband progress (default: "no")
;call_group= ; The numeric pickup groups for a channel (default: "")
diff --git a/contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.py b/contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.py
new file mode 100644
index 0000000..ad36bd9
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.py
@@ -0,0 +1,30 @@
+"""add g726_non_standard
+
+Revision ID: 28b8e71e541f
+Revises: a541e0b5e89
+Create Date: 2015-06-12 16:07:08.609628
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '28b8e71e541f'
+down_revision = 'a541e0b5e89'
+
+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('g726_non_standard', yesno_values))
+
+
+def downgrade():
+ op.drop_column('ps_endpoints', 'g726_non_standard')
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 08d6954..5267603 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -557,6 +557,8 @@
unsigned int tos_video;
/*! Priority for video streams */
unsigned int cos_video;
+ /*! Is g.726 packed in a non standard way */
+ unsigned int g726_non_standard;
};
/*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index f90b475..8d5adf6 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -471,6 +471,15 @@
set to <literal>sdes</literal> or <literal>dtls</literal>.
</para></description>
</configOption>
+ <configOption name="g726_non_standard" default="no">
+ <synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
+ <description><para>
+ When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
+ packing order instead of what is recommended by RFC3551. Since this essentially
+ replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
+ specified in the endpoint's allowed codec list.
+ </para></description>
+ </configOption>
<configOption name="inband_progress" default="no">
<synopsis>Determines whether chan_pjsip will indicate ringing using inband
progress.</synopsis>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 4ce7735..d4fa152 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1923,6 +1923,7 @@
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "dtls_fingerprint", "", dtls_handler, dtlsfingerprint_to_str, NULL, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "srtp_tag_32", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.srtp_tag_32));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "media_encryption_optimistic", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.encryption_optimistic));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "g726_non_standard", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.g726_non_standard));
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "redirect_method", "user", redirect_handler, NULL, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "set_var", "", set_var_handler, set_var_to_str, set_var_to_vl, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "message_context", "", OPT_STRINGFIELD_T, 1, STRFLDSET(struct ast_sip_endpoint, message_context));
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 3f48683..22c4529 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -155,6 +155,8 @@
char name[256];
char media[20];
char fmt_param[256];
+ enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
+ AST_RTP_OPT_G726_NONSTANDARD : 0;
ast_rtp_codecs_payloads_initialize(codecs);
@@ -176,9 +178,10 @@
if (strcmp(name,"telephone-event") == 0) {
tel_event++;
}
+
ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]),
- media, name, 0, rtpmap->clock_rate);
+ media, name, options, rtpmap->clock_rate);
/* Look for an optional associated fmtp attribute */
if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) {
continue;
@@ -304,18 +307,20 @@
return 0;
}
-static pjmedia_sdp_attr* generate_rtpmap_attr(pjmedia_sdp_media *media, pj_pool_t *pool, int rtp_code,
- int asterisk_format, struct ast_format *format, int code)
+static pjmedia_sdp_attr* generate_rtpmap_attr(struct ast_sip_session *session, pjmedia_sdp_media *media, pj_pool_t *pool,
+ int rtp_code, int asterisk_format, struct ast_format *format, int code)
{
pjmedia_sdp_rtpmap rtpmap;
pjmedia_sdp_attr *attr = NULL;
char tmp[64];
+ enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
+ AST_RTP_OPT_G726_NONSTANDARD : 0;
snprintf(tmp, sizeof(tmp), "%d", rtp_code);
pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1];
rtpmap.clock_rate = ast_rtp_lookup_sample_rate2(asterisk_format, format, code);
- pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, 0));
+ pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options));
rtpmap.param.slen = 0;
rtpmap.param.ptr = NULL;
@@ -1051,7 +1056,7 @@
continue;
}
- if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 1, format, 0))) {
+ if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
ao2_ref(format, -1);
continue;
}
@@ -1076,7 +1081,7 @@
continue;
}
- if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 0, NULL, index))) {
+ if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 0, NULL, index))) {
continue;
}
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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