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

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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
This commit is contained in:
Kevin Harwell
2015-06-12 16:58:27 -05:00
parent 2618d1e638
commit 31c77b157b
7 changed files with 64 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

17
res/res_pjsip_sdp_rtp.c
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@@ -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;
}