static void apply_packetization(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *remote_stream)
{
pjmedia_sdp_attr *attr;
pj_str_t value;
unsigned long framing;
int codec;
struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(session_media->rtp)->pref;
/* Apply packetization if available and configured to do so */
if (!session->endpoint->media.rtp.use_ptime || !(attr = pjmedia_sdp_media_find_attr2(remote_stream, "ptime", NULL))) {
return;
}
value = attr->value;
framing = pj_strtoul(pj_strltrim(&value));
for (codec = 0; codec < AST_RTP_MAX_PT; codec++) {
struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(
session_media->rtp), codec);
if (!format.asterisk_format) {
continue;
}
ast_codec_pref_setsize(pref, &format.format, framing);
}
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp, pref);
}
/*! \brief Internal function which creates an RTP instance */
static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
{
struct ast_rtp_engine_ice *ice;
if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->media.rtp.engine);
return -1;
}
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->media.rtp.symmetric);
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp, &session->endpoint->media.prefs);
if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
if (!session->endpoint->media.rtp.ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
ice->stop(session_media->rtp);
}
if (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
} else if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
if (!strcmp(session_media->stream_type, STR_AUDIO) &&
(session->endpoint->media.tos_audio || session->endpoint->media.cos_video)) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
session->endpoint->media.cos_audio, "SIP RTP Audio");
} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&
(session->endpoint->media.tos_video || session->endpoint->media.cos_video)) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
session->endpoint->media.cos_video, "SIP RTP Video");
}
return 0;
}
static int set_caps(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *stream)
{
RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, peer, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, joint, NULL, ao2_cleanup);
enum ast_media_type media_type = stream_to_media_type(session_media->stream_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;
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", session_media->stream_type);
return -1;
}
/* get the endpoint capabilities */
if (direct_media_enabled) {
ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
format_cap_only_type(caps, media_type);
} else {
ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
}
/* get the capabilities on the peer */
get_codecs(session, stream, &codecs, session_media);
ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
/* get the joint capabilities between peer and endpoint */
ast_format_cap_get_compatible(caps, peer, joint);
if (!ast_format_cap_count(joint)) {
struct ast_str *usbuf = ast_str_alloca(256);
struct ast_str *thembuf = ast_str_alloca(256);
ast_rtp_codecs_payloads_destroy(&codecs);
ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
session_media->stream_type,
ast_format_cap_get_names(caps, &usbuf),
ast_format_cap_get_names(peer, &thembuf));
return -1;
}
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
if (session->channel) {
ast_channel_lock(session->channel);
ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, media_type);
ast_format_cap_append_from_cap(caps, joint, media_type);
/*
* Apply the new formats to the channel, potentially changing
* raw read/write formats and translation path while doing so.
*/
ast_channel_nativeformats_set(session->channel, caps);
if (media_type == AST_MEDIA_TYPE_AUDIO) {
ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
}
if ((session->endpoint->dtmf == AST_SIP_DTMF_AUTO)
&& (ast_rtp_instance_dtmf_mode_get(session_media->rtp) == AST_RTP_DTMF_MODE_RFC2833)
&& (session->dsp)) {
dsp_features = ast_dsp_get_features(session->dsp);
dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
if (dsp_features) {
ast_dsp_set_features(session->dsp, dsp_features);
} else {
ast_dsp_free(session->dsp);
session->dsp = NULL;
}
}
ast_channel_unlock(session->channel);
}
ast_rtp_codecs_payloads_destroy(&codecs);
return 0;
}
static int set_caps(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *stream)
{
RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, peer, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, joint, NULL, ao2_cleanup);
enum ast_media_type media_type = stream_to_media_type(session_media->stream_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);
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
!(joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", session_media->stream_type);
return -1;
}
/* get the endpoint capabilities */
if (direct_media_enabled) {
ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
format_cap_only_type(caps, media_type);
} else {
ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
}
/* get the capabilities on the peer */
get_codecs(session, stream, &codecs);
ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
/* get the joint capabilities between peer and endpoint */
ast_format_cap_get_compatible(caps, peer, joint);
if (!ast_format_cap_count(joint)) {
struct ast_str *usbuf = ast_str_alloca(64);
struct ast_str *thembuf = ast_str_alloca(64);
ast_rtp_codecs_payloads_destroy(&codecs);
ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
session_media->stream_type,
ast_format_cap_get_names(caps, &usbuf),
ast_format_cap_get_names(peer, &thembuf));
return -1;
}
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
if (session->channel) {
struct ast_format *fmt;
ast_channel_lock(session->channel);
ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel), AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, media_type);
/*
* XXX Historically we picked the "best" joint format to use
* and stuck with it. It would be nice to just append the
* determined joint media capabilities to give translation
* more formats to choose from when necessary. Unfortunately,
* there are some areas of the system where this doesn't work
* very well. (The softmix bridge in particular is reluctant
* to pick higher fidelity formats and has a problem with
* asymmetric sample rates.)
*/
fmt = ast_format_cap_get_format(joint, 0);
ast_format_cap_append(caps, fmt, 0);
/*
* Apply the new formats to the channel, potentially changing
* raw read/write formats and translation path while doing so.
*/
ast_channel_nativeformats_set(session->channel, caps);
ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
ast_channel_unlock(session->channel);
ao2_ref(fmt, -1);
}
ast_rtp_codecs_payloads_destroy(&codecs);
return 0;
}
static int set_caps(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *stream)
{
RAII_VAR(struct ast_format_cap *, caps, NULL, ast_format_cap_destroy);
RAII_VAR(struct ast_format_cap *, peer, NULL, ast_format_cap_destroy);
RAII_VAR(struct ast_format_cap *, joint, NULL, ast_format_cap_destroy);
enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
struct ast_rtp_codecs codecs;
struct ast_format fmt;
int fmts = 0;
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
!ast_format_cap_is_empty(session->direct_media_cap);
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK)) ||
!(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK))) {
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", session_media->stream_type);
return -1;
}
/* get the endpoint capabilities */
if (direct_media_enabled) {
ast_format_cap_joint_copy(session->endpoint->media.codecs, session->direct_media_cap, caps);
} else {
ast_format_cap_copy(caps, session->endpoint->media.codecs);
}
format_cap_only_type(caps, media_type);
/* get the capabilities on the peer */
get_codecs(session, stream, &codecs);
ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
/* get the joint capabilities between peer and endpoint */
if (!(joint = ast_format_cap_joint(caps, peer))) {
char usbuf[64], thembuf[64];
ast_rtp_codecs_payloads_destroy(&codecs);
ast_getformatname_multiple(usbuf, sizeof(usbuf), caps);
ast_getformatname_multiple(thembuf, sizeof(thembuf), peer);
ast_log(LOG_WARNING, "No joint capabilities between our configuration(%s) and incoming SDP(%s)\n", usbuf, thembuf);
return -1;
}
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
ast_format_cap_copy(caps, session->req_caps);
ast_format_cap_remove_bytype(caps, media_type);
ast_format_cap_append(caps, joint);
ast_format_cap_append(session->req_caps, caps);
if (session->channel) {
ast_format_cap_copy(caps, ast_channel_nativeformats(session->channel));
ast_format_cap_remove_bytype(caps, media_type);
ast_codec_choose(&session->endpoint->media.prefs, joint, 1, &fmt);
ast_format_cap_add(caps, &fmt);
/* Apply the new formats to the channel, potentially changing read/write formats while doing so */
ast_format_cap_copy(ast_channel_nativeformats(session->channel), caps);
ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
}
ast_rtp_codecs_payloads_destroy(&codecs);
return 0;
}
