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add fmtp to sip 

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@1882 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Anthony Minessale 2006-07-14 19:20:54 +00:00
parent a7294a1c95
commit 2b4ddd0576
1 changed files with 22 additions and 8 deletions
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30
src/mod/endpoints/mod_exosip/mod_exosip.c
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@ -352,8 +352,10 @@ static switch_status_t exosip_on_init(switch_core_session_t *session)
sprintf(dbuf, "%u", tech_pvt->te);
sdp_message_m_payload_add(tech_pvt->local_sdp, 0, osip_strdup(dbuf));
sdp_add_codec(tech_pvt->sdp_config, SWITCH_CODEC_TYPE_AUDIO, tech_pvt->te, "telephone-event", 8000, 0);
sprintf(dbuf, "%u telephone-event/8000\na=fmtp %u 0-15", tech_pvt->te, tech_pvt->te);
sprintf(dbuf, "%u telephone-event/8000", tech_pvt->te);
sdp_message_a_attribute_add(tech_pvt->local_sdp, 0, "rtpmap", osip_strdup(dbuf));
sprintf(dbuf, "%u 0-15", tech_pvt->te);
sdp_message_a_attribute_add(tech_pvt->local_sdp, 0, "fmtp", osip_strdup(dbuf));
if (tech_pvt->num_codecs > 0) {
int i, lastcode = -1;
@ -1306,11 +1308,6 @@ static switch_status_t exosip_create_call(eXosip_event_t * event)
sdp_message_init(&tech_pvt->local_sdp);
sprintf(dbuf, "%u", tech_pvt->te);
sdp_message_m_payload_add(tech_pvt->local_sdp, 0, osip_strdup(dbuf));
sdp_add_codec(tech_pvt->sdp_config, SWITCH_CODEC_TYPE_AUDIO, tech_pvt->te, "telephone-event", 8000, 0);
sprintf(dbuf, "%u telephone-event/8000\na=fmtp %u 0-15", tech_pvt->te, tech_pvt->te);
sdp_message_a_attribute_add(tech_pvt->local_sdp, 0, "rtpmap", osip_strdup(dbuf));
if (tech_pvt->num_codecs > 0) {
int i;
@ -1318,8 +1315,12 @@ static switch_status_t exosip_create_call(eXosip_event_t * event)
for (i = 0; i < tech_pvt->num_codecs; i++) {
for (imp = tech_pvt->codecs[i]; imp; imp = imp->next) {
sdp_add_codec(tech_pvt->sdp_config, tech_pvt->codecs[i]->codec_type, imp->ianacode, imp->iananame,
imp->samples_per_second, 0);
sdp_add_codec(tech_pvt->sdp_config,
tech_pvt->codecs[i]->codec_type,
imp->ianacode,
imp->iananame,
imp->samples_per_second,
0);
}
}
@ -1335,6 +1336,7 @@ static switch_status_t exosip_create_call(eXosip_event_t * event)
sdp_message_to_str(remote_sdp, &remote_sdp_str);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "REMOTE SDP:\n%s", remote_sdp_str);
mline = 0;
while (0 == osip_rfc3264_match(tech_pvt->sdp_config, remote_sdp, audio_tab, video_tab, t38_tab, app_tab, mline)) {
if (audio_tab[0] == NULL && video_tab[0] == NULL && t38_tab[0] == NULL && app_tab[0] == NULL) {
@ -1355,6 +1357,18 @@ static switch_status_t exosip_create_call(eXosip_event_t * event)
done:
free(remote_sdp_str);
sprintf(dbuf, "%u", tech_pvt->te);
sdp_message_m_payload_add(tech_pvt->local_sdp, 0, osip_strdup(dbuf));
sdp_add_codec(tech_pvt->sdp_config, SWITCH_CODEC_TYPE_AUDIO, tech_pvt->te, "telephone-event", 8000, 0);
sprintf(dbuf, "%u telephone-event/8000", tech_pvt->te);
sdp_message_a_attribute_add(tech_pvt->local_sdp, 0, "rtpmap", osip_strdup(dbuf));
sprintf(dbuf, "%u 0-15", tech_pvt->te);
sdp_message_a_attribute_add(tech_pvt->local_sdp, 0, "fmtp", osip_strdup(dbuf));
sdp_message_o_origin_set(tech_pvt->local_sdp, "FreeSWITCH", "0", "0", "IN", "IP4", ip);
sdp_message_s_name_set(tech_pvt->local_sdp, "SIP Call");