[asterisk-commits] res pjsip: Add rtp keepalive to sample config file. (asterisk[13])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 24 10:42:56 CDT 2015
Joshua Colp has submitted this change and it was merged.
Change subject: res_pjsip: Add rtp_keepalive to sample config file.
......................................................................
res_pjsip: Add rtp_keepalive to sample config file.
Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19
---
M configs/samples/pjsip.conf.sample
1 file changed, 3 insertions(+), 0 deletions(-)
Approvals:
Richard Mudgett: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, approved; Verified
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 239efda..6afe053 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -732,6 +732,9 @@
; byte tags (default: "no")
;set_var= ; Variable set on a channel involving the endpoint. For multiple
; channel variables specify multiple 'set_var'(s)
+;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if
+ ; RTP is not flowing. This setting is useful for ensuring that
+ ; holes in NATs and firewalls are kept open throughout a call.
;==========================AUTH SECTION OPTIONS=========================
;[auth]
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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