diff options
| author | Mark Spencer <markster@digium.com> | 2006-05-09 11:44:50 +0000 |
|---|---|---|
| committer | Mark Spencer <markster@digium.com> | 2006-05-09 11:44:50 +0000 |
| commit | 9953f4f40eddceccd4d99f07ffe9e8a380e371e9 (patch) | |
| tree | d57b0a40dccb81bbcb68b1d0c7e720f6a331ffbb /channels | |
| parent | 66dda468d191ced212eaa144fea98323aacf4a24 (diff) | |
Make SIP early media work more efficiently without so many reinvites
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 19 |
1 files changed, 13 insertions, 6 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 4d4a3e1ef..5881d2d96 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -13588,6 +13588,7 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan) static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active) { struct sip_pvt *p; + int changed = 0; p = chan->tech_pvt; if (!p) @@ -13598,17 +13599,23 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc ast_mutex_unlock(&p->lock); return 0; } - if (rtp) - ast_rtp_get_peer(rtp, &p->redirip); + if (rtp) + changed |= ast_rtp_get_peer(rtp, &p->redirip); else memset(&p->redirip, 0, sizeof(p->redirip)); if (vrtp) - ast_rtp_get_peer(vrtp, &p->vredirip); + changed |= ast_rtp_get_peer(vrtp, &p->vredirip); else memset(&p->vredirip, 0, sizeof(p->vredirip)); - p->redircodecs = codecs; - if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { - if (!p->pendinginvite) { + if (p->redircodecs != codecs) { + p->redircodecs = codecs; + changed = 1; + } + if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) { + if (chan->_state != AST_STATE_UP) { + char iabuf[INET_ADDRSTRLEN]; + ast_log(LOG_DEBUG, "Early media setting SIP '%s' - Sending early media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip)); + } else if (!p->pendinginvite) { if (option_debug > 2) { char iabuf[INET_ADDRSTRLEN]; ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip)); |