/*! \brief Function which negotiates an incoming media stream */
static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
{
char host[NI_MAXHOST];
RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
enum ast_format_type media_type = stream_to_media_type(session_media->stream_type);
/* If no type formats have been configured reject this stream */
if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type)) {
return 0;
}
/* Ensure incoming transport is compatible with the endpoint's configuration */
if (!session->endpoint->media.rtp.use_received_transport &&
check_endpoint_media_transport(session->endpoint, stream) == AST_SIP_MEDIA_TRANSPORT_INVALID) {
return -1;
}
ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
/* The provided host was actually invalid so we error out this negotiation */
return -1;
}
/* Using the connection information create an appropriate RTP instance */
if (!session_media->rtp && create_rtp(session, session_media, ast_sockaddr_is_ipv6(addrs))) {
return -1;
}
if (session->endpoint->media.rtp.use_received_transport) {
pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport);
}
if (setup_media_encryption(session, session_media, stream)) {
return -1;
}
if (set_caps(session, session_media, stream)) {
return -1;
}
if (media_type == AST_FORMAT_TYPE_AUDIO) {
apply_packetization(session, session_media, stream);
}
return 1;
}
/*! \brief Function which negotiates an incoming media stream */
static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
{
char host[NI_MAXHOST];
RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free);
enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
enum ast_sip_session_media_encryption encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
int res;
/* If port is 0, ignore this media stream */
if (!stream->desc.port) {
ast_debug(3, "Media stream '%s' is already declined\n", session_media->stream_type);
return 0;
}
/* If no type formats have been configured reject this stream */
if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type)) {
ast_debug(3, "Endpoint has no codecs for media type '%s', declining stream\n", session_media->stream_type);
return 0;
}
/* Ensure incoming transport is compatible with the endpoint's configuration */
if (!session->endpoint->media.rtp.use_received_transport) {
encryption = check_endpoint_media_transport(session->endpoint, stream);
if (encryption == AST_SIP_MEDIA_TRANSPORT_INVALID) {
return -1;
}
}
ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
/* Ensure that the address provided is valid */
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
/* The provided host was actually invalid so we error out this negotiation */
return -1;
}
/* Using the connection information create an appropriate RTP instance */
if (!session_media->rtp && create_rtp(session, session_media, ast_sockaddr_is_ipv6(addrs))) {
return -1;
}
res = setup_media_encryption(session, session_media, sdp, stream);
if (res) {
if (!session->endpoint->media.rtp.encryption_optimistic) {
/* If optimistic encryption is disabled and crypto should have been enabled
* but was not this session must fail.
*/
return -1;
}
/* There is no encryption, sad. */
session_media->encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
}
/* If we've been explicitly configured to use the received transport OR if
* encryption is on and crypto is present use the received transport.
* This is done in case of optimistic because it may come in as RTP/AVP or RTP/SAVP depending
* on the configuration of the remote endpoint (optimistic themselves or mandatory).
*/
if ((session->endpoint->media.rtp.use_received_transport) ||
((encryption == AST_SIP_MEDIA_ENCRYPT_SDES) && !res)) {
pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport);
}
if (set_caps(session, session_media, stream)) {
return 0;
}
return 1;
}
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;
}
