diff --git a/CHANGES b/CHANGES
index 281d059e46d432f0c6327f8e8542975100680eb4..d2fa84c025d7fb3410342032fc4c9bfb6a2bf43f 100644
--- a/CHANGES
+++ b/CHANGES
@@ -186,6 +186,12 @@ AMI
* 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 276e214e9d5e3d66b128c296a4d8c1b6221f5779..24ff327f856af6a963347ce7a5062b24992164ee 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 0000000000000000000000000000000000000000..ad36bd9b7cd3c3b87271caf34532a8ba03687832
--- /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 08d695411c45cb062c19e7d7f35e0108f0e0683c..5267603fab1782e23928a7ba41d3b5374779cf94 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -557,6 +557,8 @@ struct ast_sip_endpoint_media_configuration {
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 f90b47552fe486feb9d0ca97179f630611667194..8d5adf6a7f38b486a7f65a62bdd826dd5c5b808e 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 4ce773563115ecb2b1e34da53f903f1925dc8445..d4fa1521b95347a08057e091f0fb4897da924422 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1923,6 +1923,7 @@ int ast_res_pjsip_initialize_configuration(void)
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 3f48683511cea66919521f8ae066fba919afa3d4..22c4529d9e80d58a8378f312ee872e96379c63f9 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -155,6 +155,8 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
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 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
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 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
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 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
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 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
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;
}