[Asterisk-code-review] res pjsip: Re-use IP version of signaling (SIP) for media (R... (asterisk[13])

[George Joseph]

George Joseph has posted comments on this change.

Change subject: res_pjsip: Re-use IP version of signaling (SIP) for media (RTP).
......................................................................


Patch Set 2:

(1 comment)

https://gerrit.asterisk.org/#/c/3663/2/res/res_pjsip_sdp_rtp.c
File res/res_pjsip_sdp_rtp.c:

PS2, Line 1060: 		unsigned int ip6;
              : 
              : 		if (session->contact) {
              : 			/* IP6 addresses contain at least one colon. IP4 do not, because
              : 			 * a port is stored in another variable: contact->via_port
              : 			 */
              : 			ip6 = strstr(session->contact->via_addr, ":") ? 1 : 0;
              : 		} else {
              : 			ast_log(LOG_ERROR, "Cannot determine whether to use IP4 or IP6; falling back to rtp_ipv6 of endpoint. Please, report as issue!\n");
              : 			ip6 = session->endpoint->media.rtp.ipv6;
              : 		}
This code block is duplicated in a bunch of places.  You might want to make it an inline in res_pjsip.h instead.


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Gerrit-MessageType: comment
Gerrit-Change-Id: I01a85a8c6723fcc12e86139f80e090e2078d04bb
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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