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

3 files changed, 21 insertions, 6 deletions

diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index f90b47552..8d5adf6a7 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 4ce773563..d4fa1521b 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 3f4868351..22c4529d9 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;
 			}