[Asterisk-code-review] res pjsip: Add rtp keepalive to sample config file. (asterisk[13])

[Mark Michelson]

Mark Michelson has uploaded a new change for review.

  https://gerrit.asterisk.org/958

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(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/58/958/1

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: newchange
Gerrit-Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>