Changeset 2241 for pjproject/trunk/pjsip/src/pjsua-lib/pjsua_media.c
- Timestamp:
- Aug 26, 2008 4:51:28 PM (16 years ago)
- File:
-
- 1 edited
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pjproject/trunk/pjsip/src/pjsua-lib/pjsua_media.c
r2235 r2241 1159 1159 1160 1160 if (call->session) { 1161 pjmedia_rtcp_stat stat; 1162 1163 if (pjmedia_session_get_stream_stat(call->session, 1164 call->audio_idx, 1165 &stat) == PJ_SUCCESS) 1166 { 1167 /* Save RTP timestamp & sequence, so when media session is 1168 * restarted, those values will be restored as the initial 1169 * RTP timestamp & sequence of the new media session. So in 1170 * the same call session, RTP timestamp and sequence are 1171 * guaranteed to be contigue. 1172 */ 1173 call->rtp_tx_seq_ts_set = 1 | (1 << 1); 1174 call->rtp_tx_seq = stat.rtp_tx_last_seq; 1175 call->rtp_tx_ts = stat.rtp_tx_last_ts; 1176 } 1177 1161 1178 if (pjsua_var.ua_cfg.cb.on_stream_destroyed) { 1162 1179 pjsua_var.ua_cfg.cb.on_stream_destroyed(call_id, call->session, 0); … … 1322 1339 /* Set SSRC */ 1323 1340 si->ssrc = call->ssrc; 1341 1342 /* Set RTP timestamp & sequence, normally these value are intialized 1343 * automatically when stream session created, but for some cases (e.g: 1344 * call reinvite, call update) timestamp and sequence need to be kept 1345 * contigue. 1346 */ 1347 si->rtp_ts = call->rtp_tx_ts; 1348 si->rtp_seq = call->rtp_tx_seq; 1349 si->rtp_seq_ts_set = call->rtp_tx_seq_ts_set; 1324 1350 1325 1351 /* Create session based on session info. */
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