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authorMark Michelson <mmichelson@digium.com>2015-07-09 14:17:53 -0500
committerMark Michelson <mmichelson@digium.com>2015-07-20 12:37:01 -0500
commit2b42264e66656f6ab6bc664eec4e93d353c58ffe (patch)
tree86f14c98fb53e3231beacb9af3670c674612daaf /res/res_rtp_asterisk.c
parent4a25d55416c736431620ce357f2bd9e241d62372 (diff)
res_pjsip: Add rtp_keepalive endpoint option.
This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d
Diffstat (limited to 'res/res_rtp_asterisk.c')
-rw-r--r--res/res_rtp_asterisk.c7
1 files changed, 6 insertions, 1 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 0a68a2db7..53e9b29c2 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -2166,6 +2166,7 @@ static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t siz
void *temp = buf;
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
+ int res;
*ice = 0;
@@ -2184,7 +2185,11 @@ static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t siz
}
#endif
- return ast_sendto(rtcp ? rtp->rtcp->s : rtp->s, temp, len, flags, sa);
+ res = ast_sendto(rtcp ? rtp->rtcp->s : rtp->s, temp, len, flags, sa);
+ if (res > 0) {
+ ast_rtp_instance_set_last_tx(instance, time(NULL));
+ }
+ return res;
}
static int rtcp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)