Author: rizzo
Date: Fri Jul 20 21:34:34 2007
New Revision: 76230
URL: http://svn.digium.com/view/asterisk?view=rev&rev=76230
Log:
more merge from trunk: one piece of code related to g723, whitespace fixes,
and remove an unnecessary null pointer check before ast_strlen_zero
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL:
http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=76230&r1=76229&r2=76230
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Fri Jul 20 21:34:34 2007
@@ -6788,6 +6788,9 @@
if (codec == AST_FORMAT_G729A) {
/* Indicate that we don't support VAD (G.729 annex B) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
+ } else if (codec == AST_FORMAT_G723_1) {
+ /* Indicate that we don't support VAD (G.723.1 annex A) */
+ ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
} else if (codec == AST_FORMAT_ILBC) {
/* Add information about us using only 20/30 ms packetization */
ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code,
fmt.cur_ms);
@@ -6795,6 +6798,7 @@
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
*min_packet_size = fmt.cur_ms;
+
/* Our first codec packetization processed cannot be zero */
if ((*min_packet_size)==0 && fmt.cur_ms)
*min_packet_size = fmt.cur_ms;
@@ -6928,7 +6932,6 @@
ast_str_append(&m_modem, 0, "s=session\r\n");
ast_str_append(&m_modem, 0, "c=IN IP4 %s\r\n",
ast_inet_ntoa(udptldest.sin_addr));
ast_str_append(&m_modem, 0, "t=0 0\r\n");
-
ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n",
ntohs(udptldest.sin_port));
if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0)
@@ -6975,8 +6978,7 @@
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype(0, format, 0),
sample_rate);
- if (format == AST_RTP_DTMF)
- /* Indicate we support DTMF and FLASH... */
+ if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */
ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}
@@ -7139,7 +7141,7 @@
}
ast_str_append(&m_text, 0, "m=text %d RTP/AVP",
ntohs(tdest.sin_port));
- if (debug)
+ if (debug) /* XXX should I use tdest below ? */
ast_verbose("Text is at %s port %d\n",
ast_inet_ntoa(p->ourip.sin_addr), ntohs(tsin.sin_port));
}
@@ -7238,6 +7240,7 @@
/* XXX don't think you can have ptime for text */
if (min_text_packet_size)
ast_str_append(&a_text, 0, "a=ptime:%d\r\n",
min_text_packet_size);
+
if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used <
2 ||
m_text->len - m_text->used < 2 || a_text->len - a_text->used <
2 ||
a_audio->len - a_audio->used < 2 || a_video->len -
a_video->used < 2)
@@ -7273,7 +7276,7 @@
add_line(resp, a_video->str);
add_line(resp, hold); /* Repeat hold for the video stream */
}
- if (needtext) {
+ if (needtext) { /* only if text response is appropriate */
add_line(resp, m_text->str);
add_line(resp, a_text->str);
add_line(resp, hold); /* Repeat hold for the text stream */
@@ -7715,7 +7718,7 @@
}
/* This new INVITE is part of an attended transfer. Make sure that the
other end knows and replace the current call with this new call */
- if (p->options && p->options->replaces &&
!ast_strlen_zero(p->options->replaces)) {
+ if (p->options && !ast_strlen_zero(p->options->replaces)) {
add_header(&req, "Replaces", p->options->replaces);
add_header(&req, "Require", "replaces");
}
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