/*! \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);
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 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
} 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;
}
/*! \brief Function called when we should prepare to call the destination */
static struct ast_channel *multicast_rtp_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
{
char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
struct ast_rtp_instance *instance;
struct ast_sockaddr control_address;
struct ast_sockaddr destination_address;
struct ast_channel *chan;
format_t fmt = ast_best_codec(format);
ast_sockaddr_setnull(&control_address);
/* If no type was given we can't do anything */
if (ast_strlen_zero(multicast_type)) {
goto failure;
}
if (!(destination = strchr(tmp, '/'))) {
goto failure;
}
*destination++ = '\0';
if ((control = strchr(destination, '/'))) {
*control++ = '\0';
if (!ast_sockaddr_parse(&control_address, control,
PARSE_PORT_REQUIRE)) {
goto failure;
}
}
if (!ast_sockaddr_parse(&destination_address, destination,
PARSE_PORT_REQUIRE)) {
goto failure;
}
if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
goto failure;
}
if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", requestor ? requestor->linkedid : "", 0, "MulticastRTP/%p", instance))) {
ast_rtp_instance_destroy(instance);
goto failure;
}
ast_rtp_instance_set_remote_address(instance, &destination_address);
chan->tech = &multicast_rtp_tech;
chan->nativeformats = fmt;
chan->writeformat = fmt;
chan->readformat = fmt;
chan->rawwriteformat = fmt;
chan->rawreadformat = fmt;
chan->tech_pvt = instance;
return chan;
failure:
*cause = AST_CAUSE_FAILURE;
return NULL;
}
/*! \brief Function called when we should prepare to call the destination */
static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
{
char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
struct ast_rtp_instance *instance;
struct ast_sockaddr control_address;
struct ast_sockaddr destination_address;
struct ast_channel *chan;
struct ast_format fmt;
ast_best_codec(cap, &fmt);
ast_sockaddr_setnull(&control_address);
/* If no type was given we can't do anything */
if (ast_strlen_zero(multicast_type)) {
goto failure;
}
if (!(destination = strchr(tmp, '/'))) {
goto failure;
}
*destination++ = '\0';
if ((control = strchr(destination, '/'))) {
*control++ = '\0';
if (!ast_sockaddr_parse(&control_address, control,
PARSE_PORT_REQUIRE)) {
goto failure;
}
}
if (!ast_sockaddr_parse(&destination_address, destination,
PARSE_PORT_REQUIRE)) {
goto failure;
}
if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
goto failure;
}
if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", requestor ? ast_channel_linkedid(requestor) : "", 0, "MulticastRTP/%p", instance))) {
ast_rtp_instance_destroy(instance);
goto failure;
}
ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
ast_rtp_instance_set_remote_address(instance, &destination_address);
ast_channel_tech_set(chan, &multicast_rtp_tech);
ast_format_cap_add(ast_channel_nativeformats(chan), &fmt);
ast_format_copy(ast_channel_writeformat(chan), &fmt);
ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
ast_format_copy(ast_channel_readformat(chan), &fmt);
ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
ast_channel_tech_pvt_set(chan, instance);
return chan;
failure:
*cause = AST_CAUSE_FAILURE;
return NULL;
}