[svn-commits] rizzo: branch rizzo/astobj2 r57977 -
/team/rizzo/astobj2/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Mar 6 01:21:54 MST 2007
Author: rizzo
Date: Tue Mar 6 02:21:53 2007
New Revision: 57977
URL: http://svn.digium.com/view/asterisk?view=rev&rev=57977
Log:
merge from trunk svn 54838, also fixing some previous errors.
Now we have up to svn54372 plus 54838
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=57977&r1=57976&r2=57977
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Tue Mar 6 02:21:53 2007
@@ -1689,7 +1689,7 @@
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active);
static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-tatic enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
+static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static int sip_get_codec(struct ast_channel *chan);
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
@@ -4854,11 +4854,11 @@
/* If the global videosupport flag is on, we always create a RTP interface for video */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- f (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT))
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT))
p->trtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
- f (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)
+ if (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)
|| (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) {
ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n",
ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "",
@@ -6696,55 +6696,56 @@
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
*min_packet_size = fmt.cur_ms;
-
-/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
- /* This is different to the audio one now so we can add more caps later */
- static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
- char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
- int debug, int *min_packet_size)
- {
- int rtp_code;
-
- if (!p->vrtp)
- return;
-
- if (debug)
- ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
-
- if ((rtp_code = ast_rtp_lookup_code(p->vrtp, 1, codec)) == -1)
- return;
-
- ast_build_string(m_buf, m_size, " %d", rtp_code);
- ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
- /* Add fmtp code here */
- }
-
- /*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
- static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
- char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
- int debug, int *min_packet_size)
- {
- int rtp_code;
-
- if (!p->trtp)
- return;
-
- if (debug)
- ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
-
- if ((rtp_code = ast_rtp_lookup_code(p->trtp, 1, codec)) == -1)
- return;
-
- ast_build_string(m_buf, m_size, " %d", rtp_code);
- ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
- /* Add fmtp code here */
- }
/* Our first codec packetization processed cannot be zero */
if ((*min_packet_size)==0 && fmt.cur_ms)
*min_packet_size = fmt.cur_ms;
}
+
+/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
+ /* This is different to the audio one now so we can add more caps later */
+static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+ struct ast_str **m_buf, struct ast_str **a_buf,
+ int debug, int *min_packet_size)
+{
+ int rtp_code;
+
+ if (!p->vrtp)
+ return;
+
+ if (debug)
+ ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+
+ if ((rtp_code = ast_rtp_lookup_code(p->vrtp, 1, codec)) == -1)
+ return;
+
+ ast_str_append(m_buf, 0, " %d", rtp_code);
+ ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
+ ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ /* Add fmtp code here */
+}
+
+/*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
+static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+ struct ast_str **m_buf, struct ast_str **a_buf,
+ int debug, int *min_packet_size)
+{
+ int rtp_code;
+
+ if (!p->trtp)
+ return;
+
+ if (debug)
+ ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+
+ if ((rtp_code = ast_rtp_lookup_code(p->trtp, 1, codec)) == -1)
+ return;
+
+ ast_str_append(m_buf, 0, " %d", rtp_code);
+ ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
+ ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ /* Add fmtp code here */
+}
+
/*! \brief Get Max T.38 Transmission rate from T38 capabilities */
static int t38_get_rate(int t38cap)
@@ -6946,15 +6947,19 @@
char *hold;
struct ast_str *m_audio = ast_str_alloca(256); /* Media declaration line for audio */
struct ast_str *m_video = ast_str_alloca(256); /* Media declaration line for video */
+ struct ast_str *m_text = ast_str_alloca(256); /* Media declaration line for text */
struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */
struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
+ struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */
int x;
int capability;
int needvideo = FALSE;
+ int needtext = FALSE;
int debug = sip_debug_test_pvt(p);
int min_audio_packet_size = 0;
int min_video_packet_size = 0;
+ int min_text_packet_size = 0;
if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
@@ -6974,7 +6979,8 @@
if (option_debug > 1) {
char codecbuf[BUFSIZ];
- ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
+ ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
+ ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False", ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT) ? "True" : "False");
ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
}
@@ -6996,7 +7002,10 @@
}
/* Get our media addresses */
- get_our_media_address(p, needvideo, &sin, &vsin, &dest, &vdest);
+ get_our_media_address(p, needvideo, &sin, &vsin, &tsin, &dest, &vdest);
+
+ if (debug)
+ ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(sin.sin_port));
/* Ok, we need video. Let's add what we need for video and set codecs.
Video is handled differently than audio since we can not transcode. */
@@ -7009,9 +7018,38 @@
if (debug)
ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(vsin.sin_port));
}
-
- if (debug)
- ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(sin.sin_port));
+ /* Check if we need text in this call */
+ if((capability & AST_FORMAT_TEXT_MASK) && !ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT)) {
+ if (sipdebug_text)
+ ast_verbose("We think we can do text\n");
+ if (p->trtp) {
+ if (sipdebug_text)
+ ast_verbose("And we have a text rtp object\n");
+ needtext = TRUE;
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "This call needs text offers! \n");
+ } else if (option_debug > 1)
+ ast_log(LOG_DEBUG, "This call needs text offers, but there's no text support enabled ! \n");
+ }
+
+ /* Ok, we need text. Let's add what we need for text and set codecs.
+ Text is handled differently than audio since we can not transcode. */
+ if (needtext) {
+ if (sipdebug_text)
+ ast_verbose("Lets set up the text sdp\n");
+ /* Determine text destination */
+ if (p->tredirip.sin_addr.s_addr) {
+ tdest.sin_addr = p->tredirip.sin_addr;
+ tdest.sin_port = p->tredirip.sin_port;
+ } else {
+ tdest.sin_addr = p->ourip;
+ tdest.sin_port = tsin.sin_port;
+ }
+ ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port));
+
+ if (debug)
+ ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(tsin.sin_port));
+ }
/* Start building generic SDP headers */
@@ -7067,7 +7105,7 @@
}
/* Now send any other common audio and video codecs, and non-codec formats: */
- for (x = 1; x <= (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
+ for (x = 1; x <= (needtext ? AST_FORMAT_MAX_TEXT : (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO)); x <<= 1) {
if (!(capability & x)) /* Codec not requested */
continue;
@@ -7077,9 +7115,12 @@
if (x <= AST_FORMAT_MAX_AUDIO)
add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
&m_audio, &a_audio, debug, &min_audio_packet_size);
- else
- add_codec_to_sdp(p, x, 90000,
+ else if (x <= AST_FORMAT_MAX_VIDEO)
+ add_vcodec_to_sdp(p, x, 90000,
&m_video, &a_video, debug, &min_video_packet_size);
+ else if (x <= AST_FORMAT_MAX_TEXT)
+ add_tcodec_to_sdp(p, x, 1000,
+ &m_text, &a_text, debug, &min_text_packet_size);
}
/* Now add DTMF RFC2833 telephony-event as a codec */
@@ -7098,21 +7139,31 @@
if (min_audio_packet_size)
ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
+
+ /* XXX don't think you can have ptime for video */
if (min_video_packet_size)
ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);
+ /* 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)
ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
ast_str_append(&m_audio, 0, "\r\n");
if (needvideo)
ast_str_append(&m_video, 0, "\r\n");
+ if (needtext)
+ ast_str_append(&m_text, 0, "\r\n");
len = strlen(version) + strlen(subject) + strlen(owner) +
strlen(connection) + strlen(stime) + m_audio->used + a_audio->used + strlen(hold);
if (needvideo) /* only if video response is appropriate */
len += m_video->used + a_video->used + strlen(bandwidth) + strlen(hold);
+ if (needtext) /* only if text response is appropriate */
+ len += m_text->used + a_text->used + strlen(hold);
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
@@ -7130,6 +7181,11 @@
add_line(resp, m_video->str);
add_line(resp, a_video->str);
add_line(resp, hold); /* Repeat hold for the video stream */
+ }
+ if (needtext) {
+ add_line(resp, m_text->str);
+ add_line(resp, a_text->str);
+ add_line(resp, hold); /* Repeat hold for the text stream */
}
/* Update lastrtprx when we send our SDP */
@@ -9922,6 +9978,11 @@
ast_rtp_destroy(p->vrtp);
p->vrtp = NULL;
}
+ /* If we do not support text, remove text from call structure */
+ if (!ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && p->trtp) {
+ ast_rtp_destroy(p->trtp);
+ p->trtp = NULL;
+ }
}
unref_user(user);
return res;
@@ -10041,6 +10102,10 @@
ast_rtp_destroy(p->vrtp);
p->vrtp = NULL;
}
+ if ((!ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) || !(p->capability & AST_FORMAT_TEXT_MASK)) && p->trtp) {
+ ast_rtp_destroy(p->trtp);
+ p->trtp = NULL;
+ }
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF;
@@ -10527,6 +10592,7 @@
"Dynamic: %s\r\n"
"Natsupport: %s\r\n"
"VideoSupport: %s\r\n"
+ "TextSupport: %s\r\n"
"ACL: %s\r\n"
"Status: %s\r\n"
"RegisterFrom: %s\r\n"
@@ -10538,6 +10604,7 @@
ast_test_flag(&iterator->flags[1], SIP_PAGE2_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */
ast_test_flag(&iterator->flags[0], SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */
ast_test_flag(&iterator->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */
+ ast_test_flag(&iterator->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no", /* TEXTSUPPORT=yes? */
iterator->ha ? "yes" : "no", /* permit/deny */
status,
iterator->register_from_hdr,
@@ -10976,6 +11043,7 @@
ast_cli(fd, " PromiscRedir : %s\n", ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Yes":"No");
ast_cli(fd, " User=Phone : %s\n", ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Yes":"No");
ast_cli(fd, " Video Support: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Yes":"No");
+ ast_cli(fd, " Text Support : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Yes":"No");
ast_cli(fd, " Trust RPID : %s\n", ast_test_flag(&peer->flags[0], SIP_TRUSTRPID) ? "Yes" : "No");
ast_cli(fd, " Send RPID : %s\n", ast_test_flag(&peer->flags[0], SIP_SENDRPID) ? "Yes" : "No");
ast_cli(fd, " Subscriptions: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
@@ -11065,6 +11133,7 @@
astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
+ astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -11214,6 +11283,7 @@
ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port));
ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr));
ast_cli(fd, " Videosupport: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "Yes" : "No");
+ ast_cli(fd, " Textsupport: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT) ? "Yes" : "No");
ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No");
ast_cli(fd, " MatchAuthUsername: %s\n", global_match_auth_username ? "Yes" : "No");
ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No");
@@ -11238,6 +11308,7 @@
ast_cli(fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip));
ast_cli(fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
ast_cli(fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
+ ast_cli(fd, " IP ToS RTP text: %s\n", ast_tos2str(global_tos_text));
ast_cli(fd, " T38 fax pt UDPTL: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL) ? "Yes" : "No");
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
ast_cli(fd, " T38 fax pt RTP: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP) ? "Yes" : "No");
@@ -11909,11 +11980,13 @@
if (a->argc == e->args) { /* on/off */
if (!strcasecmp(what, "on")) {
ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE);
+ ast_set_flag(&global_flags[1], SIP_PAGE2_DEBUG_TEXT); /*! \note this can be a special debug command - "sip debug text" or something */
memset(&debugaddr, 0, sizeof(debugaddr));
ast_cli(a->fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : "");
return CLI_SUCCESS;
} else if (!strcasecmp(what, "off")) {
ast_clear_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE);
+ ast_clear_flag(&global_flags[1], SIP_PAGE2_DEBUG_TEXT);
ast_cli(a->fd, "SIP Debugging Disabled\n");
return CLI_SUCCESS;
}
@@ -13161,6 +13234,8 @@
ast_rtp_stop(p->rtp);
if (p->vrtp)
ast_rtp_stop(p->vrtp);
+ if (p->trtp)
+ ast_rtp_stop(p->trtp);
if (p->udptl)
ast_udptl_stop(p->udptl);
}
@@ -15174,7 +15249,7 @@
/* Get RTCP quality before end of call */
if (record_history(p) || p->owner) {
- char *audioqos, *videoqos;
+ char *audioqos, *videoqos, *textqos;
if (p->rtp) {
audioqos = ast_rtp_get_quality(p->rtp);
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
@@ -15186,6 +15261,12 @@
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
+ }
+ if (p->trtp) {
+ textqos = ast_rtp_get_quality(p->trtp);
+ append_history(p, "RTCPtext", "Quality:%s", textqos);
+ if (p->owner)
+ pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", textqos);
}
}
@@ -16474,6 +16555,7 @@
printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif
p->prefcodec = oldformat; /* Format for this call */
+ p->jointcapability = oldformat;
sip_pvt_lock(p);
tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */
/* XXX who keeps the reference ? */
@@ -16658,6 +16740,8 @@
CO_SF(0, SIP_PROMISCREDIR);
} else if (!strcasecmp(v->name, "videosupport")) {
CO_SF(1, SIP_PAGE2_VIDEOSUPPORT);
+ } else if (!strcasecmp(v->name, "textsupport")) {
+ CO_SF(1, SIP_PAGE2_TEXTSUPPORT);
} else if (!strcasecmp(v->name, "allowoverlap")) {
CO_SF(1, SIP_PAGE2_ALLOWOVERLAP);
} else if (!strcasecmp(v->name, "allowsubscribe")) {
@@ -17078,10 +17162,9 @@
M_STR("fromdomain", peer->fromdomain)
M_F("usereqphone", ast_set2_flag(&peer->flags[0], ast_true(v->value), SIP_USEREQPHONE);)
M_STR("fromuser", peer->fromuser)
-#if 0 /* XXX fixme */
+ /* XXX do not use commas in the macro, cpp can get confused */
M_F("outboundproxy", {
- char *port, *next, *force, *proxyname;
- int forceopt = FALSE;
+ char *port; char *next; char *force; char *proxyname; int forceopt = FALSE;
/* Set peer channel variable */
/* XXX use strsep here... */
next = proxyname = ast_strdupa(v->value);
@@ -17096,7 +17179,7 @@
/* Allocate proxy object */
peer->outboundproxy = proxy_allocate(proxyname, port, forceopt);
} )
- M_F("host", {{
+ M_F("host",
if (!strcasecmp(v->value, "dynamic")) {
/* They'll register with us */
ast_set_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC);
@@ -17124,8 +17207,8 @@
ast_copy_string(peer->tohost, v->value, sizeof(peer->tohost));
if (!peer->addr.sin_port)
peer->addr.sin_port = htons(STANDARD_SIP_PORT);
- }} )
-#endif
+ }
+ )
M_F("defaultip", {
if (ast_get_ip(&peer->defaddr, v->value)) {
unref_peer(peer);
@@ -17265,6 +17348,7 @@
global_tos_sip = DEFAULT_TOS_SIP;
global_tos_audio = DEFAULT_TOS_AUDIO;
global_tos_video = DEFAULT_TOS_VIDEO;
+ global_tos_text = DEFAULT_TOS_TEXT;
externhost[0] = '\0'; /* External host name (for behind NAT DynDNS support) */
externexpire = 0; /* Expiration for DNS re-issuing */
externrefresh = 10;
@@ -17329,6 +17413,7 @@
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT);
+ ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
/* Read the [general] config section of sip.conf (or from realtime config) */
for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
@@ -17475,6 +17560,9 @@
M_F("tos_video", {
if (ast_str2tos(v->value, &global_tos_video))
ast_log(LOG_WARNING, "Invalid tos_video value at line %d, recommended value is 'af41'. See doc/ip-tos.txt.\n", v->lineno); } )
+ M_F("tos_text", {
+ if (ast_str2tos(v->value, &global_tos_text))
+ ast_log(LOG_WARNING, "Invalid tos_text value at line %d, recommended value is 'af41'. See doc/ip-tos.txt.\n", v->lineno); } )
M_F("bindport", {
if (sscanf(v->value, "%d", &ourport) == 1) {
bindaddr.sin_port = htons(ourport);
@@ -17874,6 +17962,31 @@
return res;
}
+/* XXX this is almost the same as for vrtp. */
+/*! \brief Returns null if we can't reinvite text (part of RTP interface) */
+static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+{
+ struct sip_pvt *p = NULL;
+ enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
+
+ if (!(p = chan->tech_pvt))
+ return AST_RTP_GET_FAILED;
+
+ sip_pvt_lock(p);
+ if (!(p->trtp)) {
+ sip_pvt_unlock(p);
+ return AST_RTP_GET_FAILED;
+ }
+
+ *rtp = p->trtp;
+
+ if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ res = AST_RTP_TRY_NATIVE;
+
+ sip_pvt_unlock(p);
+
+ return res;
+}
/*! \brief Set the RTP peer for this call */
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
{
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