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res_pjsip_sdp_rtp: Preserve order of RTP codecs 

The ast_rtp_codecs_payloads functions do not preserve the order in which
the payloads were specified on an incoming SDP media line. This leads to
a problem with the codec negotiation functionality, as the format
capabilities of the stream are extracted from the ast_rtp_codecs. This
commit moves the ast_rtp_codec to ast_format conversion to the place
where the order is still known.

ASTERISK-28863
ASTERISK-29320

Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25
This commit is contained in:
Florentin Mayer 2021-11-16 13:32:43 +01:00 committed by Friendly Automation
parent 220e885bcc
commit 9eacfda84d
1 changed files with 15 additions and 8 deletions
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23
res/res_pjsip_sdp_rtp.c
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@ -313,7 +313,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
}
static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs,
struct ast_sip_session_media *session_media)
struct ast_sip_session_media *session_media, struct ast_format_cap *astformats)
{
pjmedia_sdp_attr *attr;
pjmedia_sdp_rtpmap *rtpmap;
@ -329,6 +329,8 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
ast_rtp_codecs_payloads_initialize(codecs);
ast_format_cap_remove_by_type(astformats, AST_MEDIA_TYPE_UNKNOWN);
/* Iterate through provided formats */
for (i = 0; i < stream->desc.fmt_count; ++i) {
/* The payload is kept as a string for things like t38 but for video it is always numerical */
@ -372,11 +374,19 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
ast_rtp_codecs_payload_replace_format(codecs, num, format_parsed);
ao2_ref(format_parsed, -1);
}
ao2_ref(format, -1);
}
}
}
/* Parsing done, now fill the ast_format_cap struct in the correct order */
for (i = 0; i < stream->desc.fmt_count; ++i) {
if ((format = ast_rtp_codecs_get_payload_format(codecs, pj_strtoul(&stream->desc.fmt[i])))) {
ast_format_cap_append(astformats, format, 0);
ao2_ref(format, -1);
}
}
if (!tel_event && (session->dtmf == AST_SIP_DTMF_AUTO)) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
@ -398,6 +408,7 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
unsigned long framing = pj_strtoul(pj_strltrim(&attr->value));
if (framing && session->endpoint->media.rtp.use_ptime) {
ast_rtp_codecs_set_framing(codecs, framing);
ast_format_cap_set_framing(astformats, framing);
}
}
@ -442,7 +453,6 @@ static struct ast_format_cap *set_incoming_call_offer_cap(
struct ast_format_cap *incoming_call_offer_cap;
struct ast_format_cap *remote;
struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
int fmts = 0;
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
@ -454,8 +464,7 @@ static struct ast_format_cap *set_incoming_call_offer_cap(
}
/* Get the peer's capabilities*/
get_codecs(session, stream, &codecs, session_media);
ast_rtp_codecs_payload_formats(&codecs, remote, &fmts);
get_codecs(session, stream, &codecs, session_media, remote);
incoming_call_offer_cap = ast_sip_session_create_joint_call_cap(
session, session_media->type, remote);
@ -493,7 +502,6 @@ static int set_caps(struct ast_sip_session *session,
RAII_VAR(struct ast_format_cap *, joint, NULL, ao2_cleanup);
enum ast_media_type media_type = session_media->type;
struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
int fmts = 0;
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
ast_format_cap_count(session->direct_media_cap);
int dsp_features = 0;
@ -516,8 +524,7 @@ static int set_caps(struct ast_sip_session *session,
}
/* get the capabilities on the peer */
get_codecs(session, stream, &codecs, session_media);
ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
get_codecs(session, stream, &codecs, session_media, peer);
/* get the joint capabilities between peer and endpoint */
ast_format_cap_get_compatible(caps, peer, joint);