Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity

res_pjsip: Add option to force G.726 to be treated as AAL2 packed.

Browse Source 
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
changes/60/660/1
Kevin Harwell 10 years ago
parent b8bc15286f
commit 93ac45d3bd
7 changed files with 64 additions and 6 deletions
Show Stats Download Patch File Download Diff File

6
CHANGES
Unescape View File

@ -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 ------------
------------------------------------------------------------------------------

5
configs/samples/pjsip.conf.sample
Unescape View File

@ -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: "")

30
contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.py
Unescape View File

@ -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')

2
include/asterisk/res_pjsip.h
Unescape View File

@ -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;
};
/*!

9
res/res_pjsip.c
Unescape View File

@ -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>

1
res/res_pjsip/pjsip_configuration.c
Unescape View File

@ -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));

17
res/res_pjsip_sdp_rtp.c
Unescape View File

@ -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;
}

Write Preview
Loading…
Cancel
Save