add fmtp to sip

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@1882 d0543943-73ff-0310-b7d9-9358b9ac24b2

diff --git a/src/mod/endpoints/mod_exosip/mod_exosip.c b/src/mod/endpoints/mod_exosip/mod_exosip.c
index 3adaa2da264025535378b78b029c363195c26dca..80ce2dc54e1f08b58555ea155e127dd082c0b8eb 100644 (file)
--- a/src/mod/endpoints/mod_exosip/mod_exosip.c
+++ b/src/mod/endpoints/mod_exosip/mod_exosip.c
@@ -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");