[Asterisk-code-review] res pjsip: Add rtp keepalive endpoint option. (asterisk[13])

[Richard Mudgett]

Richard Mudgett has posted comments on this change.

Change subject: res_pjsip: Add rtp_keepalive endpoint option.
......................................................................


Patch Set 3: Code-Review-1

(3 comments)

https://gerrit.asterisk.org/#/c/864/3/res/res_pjsip.c
File res/res_pjsip.c:

Line 794: 						be useful for situations where Asterisk is behind a NAT and must keep a hole open
...behind a NAT or firewall...


https://gerrit.asterisk.org/#/c/864/3/res/res_pjsip_sdp_rtp.c
File res/res_pjsip_sdp_rtp.c:

Line 125: 		ast_debug(1, "Not sending RTP keepalive on RTP instance %p since direct media is in use\n", rtp);
This debug message is likely a spammer so the level should be higher.


Line 132: 	ast_debug(1, "It has been %d seconds since RTP was last sent on instance %p. %sending keepalive\n",
Same here about spaming.


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Gerrit-MessageType: comment
Gerrit-Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
Gerrit-HasComments: Yes