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

7 files changed, 64 insertions, 6 deletions

diff --git a/CHANGES b/CHANGES
index bd79dbab7..d0363f7c3 100644
--- a/CHANGES
+++ b/CHANGES
@@ -17,6 +17,12 @@ 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 ef25a5729..239efda59 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -656,6 +656,11 @@
                         ; "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 000000000..ad36bd9b7
--- /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 0447eaa11..d4ab16499 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -555,6 +555,8 @@ 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 ff18a1d6d..27e3f8179 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 34bb69211..9a383a25a 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1922,6 +1922,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
 	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 1bd3b0cb3..fb70dd331 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;
 
@@ -1050,7 +1055,7 @@ 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;
 		}
@@ -1075,7 +1080,7 @@ 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;
 			}