[Asterisk-code-review] chan sip: No rtpmap for static RTP payload IDs in SDP. (asterisk[master])

[Joshua Colp]

Joshua Colp has submitted this change and it was merged.

Change subject: chan_sip: No rtpmap for static RTP payload IDs in SDP.
......................................................................


chan_sip: No rtpmap for static RTP payload IDs in SDP.

This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compactheaders=yes via the file sip.conf.

ASTERISK-25578 #close

Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044
---
M channels/chan_sip.c
1 file changed, 1 insertion(+), 1 deletion(-)

Approvals:
  Mark Michelson: Looks good to me, but someone else must approve
  Joshua Colp: Looks good to me, approved; Verified



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 19f8aa3..d44bf8a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12996,7 +12996,7 @@
 	/* Opus mandates 2 channels in rtpmap */
 	if (ast_format_cmp(format, ast_format_opus) == AST_FORMAT_CMP_EQUAL) {
 		ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u/2\r\n", rtp_code, mime, rate);
-	} else {
+	} else if ((35 <= rtp_code) || !(sip_cfg.compactheaders)) {
 		ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, mime, rate);
 	}
 

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Gerrit-MessageType: merged
Gerrit-Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>