Merge "res_pjsip: Add option to force G.726 to be treated as AAL2 packed."
This commit is contained in:
Joshua Colp
Gerrit Code Review
commit
7cdb858651
6
CHANGES
6
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 ------------
|
||||
------------------------------------------------------------------------------
|
||||
|
|
|
|||
|
|
@ -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: "")
|
||||
|
|
|
|||
|
|
@ -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')
|
||||
|
|
@ -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;
|
||||
};
|
||||
|
||||
/*!
|
||||
|
|
|
|||
|
|
@ -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>
|
||||
|
|
|
|||
|
|
@ -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));
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue