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

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index ae200a73185e64c790213a98b618714303ef79f9..10db010856899d36afbd2895b6db06fc4f1d07c1 100644 (file)
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -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);