[asterisk-commits] dvossel: branch dvossel/fixtheworld_phase1_step3 r301726 - in /team/dvossel/f...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 12 17:21:53 CST 2011
Author: dvossel
Date: Wed Jan 12 17:21:48 2011
New Revision: 301726
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=301726
Log:
conversion of chan_sip to ast_format api
Modified:
team/dvossel/fixtheworld_phase1_step3/channels/chan_local.c
team/dvossel/fixtheworld_phase1_step3/channels/chan_sip.c
team/dvossel/fixtheworld_phase1_step3/channels/sip/include/globals.h
team/dvossel/fixtheworld_phase1_step3/channels/sip/include/sip.h
team/dvossel/fixtheworld_phase1_step3/include/asterisk/format_cap.h
team/dvossel/fixtheworld_phase1_step3/main/format_cap.c
team/dvossel/fixtheworld_phase1_step3/main/rtp_engine.c
Modified: team/dvossel/fixtheworld_phase1_step3/channels/chan_local.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/fixtheworld_phase1_step3/channels/chan_local.c?view=diff&rev=301726&r1=301725&r2=301726
==============================================================================
--- team/dvossel/fixtheworld_phase1_step3/channels/chan_local.c (original)
+++ team/dvossel/fixtheworld_phase1_step3/channels/chan_local.c Wed Jan 12 17:21:48 2011
@@ -106,7 +106,7 @@
static int local_setoption(struct ast_channel *chan, int option, void *data, int datalen);
/* PBX interface structure for channel registration */
-static const struct ast_channel_tech local_tech = {
+static struct ast_channel_tech local_tech = {
.type = "Local",
.description = tdesc,
.requester = local_request,
@@ -1198,14 +1198,13 @@
/*! \brief Load module into PBX, register channel */
static int load_module(void)
{
- struct ast_channel_tech *tmp = (struct ast_channel_tech *) &local_tech;
- if (!(tmp->capabilities = ast_cap_alloc())) {
+ if (!(local_tech.capabilities = ast_cap_alloc())) {
return AST_MODULE_LOAD_FAILURE;
}
- ast_cap_add_all(tmp->capabilities);
+ ast_cap_add_all(local_tech.capabilities);
if (!(locals = ao2_container_alloc(BUCKET_SIZE, NULL, locals_cmp_cb))) {
- ast_cap_destroy(tmp->capabilities);
+ ast_cap_destroy(local_tech.capabilities);
return AST_MODULE_LOAD_FAILURE;
}
@@ -1213,7 +1212,7 @@
if (ast_channel_register(&local_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'Local'\n");
ao2_ref(locals, -1);
- ast_cap_destroy(tmp->capabilities);
+ ast_cap_destroy(local_tech.capabilities);
return AST_MODULE_LOAD_FAILURE;
}
ast_cli_register_multiple(cli_local, sizeof(cli_local) / sizeof(struct ast_cli_entry));
Modified: team/dvossel/fixtheworld_phase1_step3/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/fixtheworld_phase1_step3/channels/chan_sip.c?view=diff&rev=301726&r1=301725&r2=301726
==============================================================================
--- team/dvossel/fixtheworld_phase1_step3/channels/chan_sip.c (original)
+++ team/dvossel/fixtheworld_phase1_step3/channels/chan_sip.c Wed Jan 12 17:21:48 2011
@@ -1199,7 +1199,7 @@
in coming releases. */
/*--- PBX interface functions */
-static struct ast_channel *sip_request_call(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
+static struct ast_channel *sip_request_call(const char *type, struct ast_cap *cap, const struct ast_channel *requestor, void *data, int *cause);
static int sip_devicestate(void *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, char *dest, int timeout);
@@ -1291,7 +1291,7 @@
static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec);
static int process_sdp_a_image(const char *a, struct sip_pvt *p);
-static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
+static void add_codec_to_sdp(const struct sip_pvt *p, struct ast_format *codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
@@ -1565,7 +1565,7 @@
static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);
/*------- RTP Glue functions -------- */
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, format_t codecs, int nat_active);
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_cap *cap, int nat_active);
/*!--- SIP MWI Subscription support */
static int sip_subscribe_mwi(const char *value, int lineno);
@@ -1575,10 +1575,9 @@
static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);
/*! \brief Definition of this channel for PBX channel registration */
-const struct ast_channel_tech sip_tech = {
+struct ast_channel_tech sip_tech = {
.type = "SIP",
.description = "Session Initiation Protocol (SIP)",
- .capabilities = AST_FORMAT_AUDIO_MASK, /* all audio formats */
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = sip_request_call, /* called with chan unlocked */
.devicestate = sip_devicestate, /* called with chan unlocked (not chan-specific) */
@@ -4037,10 +4036,10 @@
switch (option) {
case AST_OPTION_FORMAT_READ:
- res = ast_rtp_instance_set_read_format(p->rtp, *(int *) data);
+ res = ast_rtp_instance_set_read_format(p->rtp, (struct ast_format *) data);
break;
case AST_OPTION_FORMAT_WRITE:
- res = ast_rtp_instance_set_write_format(p->rtp, *(int *) data);
+ res = ast_rtp_instance_set_write_format(p->rtp, (struct ast_format *) data);
break;
case AST_OPTION_MAKE_COMPATIBLE:
res = ast_rtp_instance_make_compatible(chan, p->rtp, (struct ast_channel *) data);
@@ -4384,6 +4383,9 @@
ast_cc_config_params_destroy(peer->cc_params);
ast_string_field_free_memory(peer);
+
+ peer->caps = ast_cap_destroy(peer->caps);
+ sip_cfg.caps = ast_cap_destroy(sip_cfg.caps);
}
/*! \brief Update peer data in database (if used) */
@@ -4825,7 +4827,7 @@
}
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) ||
- (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (dialog->capability & AST_FORMAT_VIDEO_MASK))) {
+ (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (ast_cap_has_type(dialog->caps, AST_FORMAT_TYPE_VIDEO)))) {
if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
return -1;
}
@@ -4886,7 +4888,7 @@
ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
ast_copy_flags(&dialog->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);
- dialog->capability = peer->capability;
+ ast_cap_append(peer->caps, dialog->caps);
dialog->prefs = peer->prefs;
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
/* t38pt_udptl was enabled in the peer and not in [general] */
@@ -5252,11 +5254,11 @@
return res;
}
p->callingpres = ast_party_id_presentation(&ast->caller.id);
- p->jointcapability = ast_rtp_instance_available_formats(p->rtp, p->capability, p->prefcodec);
+ ast_rtp_instance_available_formats(p->rtp, p->caps, p->prefcaps, p->jointcaps);
p->jointnoncodeccapability = p->noncodeccapability;
/* If there are no audio formats left to offer, punt */
- if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
+ if (!(ast_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO))) {
ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
res = -1;
} else {
@@ -5462,6 +5464,10 @@
ao2_ref(p->socket.tcptls_session, -1);
p->socket.tcptls_session = NULL;
}
+ p->caps = ast_cap_destroy(p->caps);
+ p->jointcaps = ast_cap_destroy(p->jointcaps);
+ p->peercaps = ast_cap_destroy(p->peercaps);
+ p->redircaps = ast_cap_destroy(p->redircaps);
}
/*! \brief update_call_counter: Handle call_limit for SIP devices
@@ -6024,7 +6030,9 @@
/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
static void try_suggested_sip_codec(struct sip_pvt *p)
{
- format_t fmt;
+ struct ast_format fmt = {
+ .id = 0,
+ };
const char *codec;
if (p->outgoing_call) {
@@ -6036,12 +6044,14 @@
if (!codec)
return;
- fmt = ast_getformatbyname(codec);
- if (fmt) {
+ ast_getformatbyname(codec, &fmt);
+ if (fmt.id) {
ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec);
- if (p->jointcapability & fmt) {
- p->jointcapability &= fmt;
- p->capability &= fmt;
+ if (ast_cap_iscompatible(p->jointcaps, &fmt)) {
+ ast_cap_remove_all(p->jointcaps);
+ ast_cap_add(p->jointcaps, &fmt);
+ ast_cap_remove_all(p->caps);
+ ast_cap_add(p->caps, &fmt);
} else
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
} else
@@ -6078,13 +6088,13 @@
switch (frame->frametype) {
case AST_FRAME_VOICE:
- if (!(frame->subclass.codec & ast->nativeformats)) {
- char s1[512], s2[512], s3[512];
+ if (!(ast_cap_iscompatible(ast->nativeformats, &frame->subclass.format))) {
+ char s1[512];
ast_log(LOG_WARNING, "Asked to transmit frame type %s, while native formats is %s read/write = %s/%s\n",
- ast_getformatname(frame->subclass.codec),
- ast_getformatname_multiple(s1, sizeof(s1), ast->nativeformats & AST_FORMAT_AUDIO_MASK),
- ast_getformatname_multiple(s2, sizeof(s2), ast->readformat),
- ast_getformatname_multiple(s3, sizeof(s3), ast->writeformat));
+ ast_getformatname(&frame->subclass.format),
+ ast_getformatname_multiple(s1, sizeof(s1), ast->nativeformats),
+ ast_getformatname(&ast->readformat),
+ ast_getformatname(&ast->writeformat));
return 0;
}
if (p) {
@@ -6537,15 +6547,18 @@
{
struct ast_channel *tmp;
struct ast_variable *v = NULL;
- format_t fmt;
- format_t what;
- format_t video;
- format_t text;
- format_t needvideo = 0;
+ struct ast_format fmt;
+ struct ast_cap *what = ast_cap_alloc();
+ struct ast_cap *video = NULL;
+ struct ast_cap *text = NULL;
+ int needvideo = 0;
int needtext = 0;
char buf[SIPBUFSIZE];
char *decoded_exten;
+ if (!what) {
+ return NULL;
+ }
{
const char *my_name; /* pick a good name */
@@ -6562,6 +6575,7 @@
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
sip_pvt_lock(i);
+ ast_cap_destroy(what);
return NULL;
}
ast_channel_lock(tmp);
@@ -6574,32 +6588,35 @@
/* Select our native format based on codec preference until we receive
something from another device to the contrary. */
- if (i->jointcapability) { /* The joint capabilities of us and peer */
- what = i->jointcapability;
- video = i->jointcapability & AST_FORMAT_VIDEO_MASK;
- text = i->jointcapability & AST_FORMAT_TEXT_MASK;
- } else if (i->capability) { /* Our configured capability for this peer */
- what = i->capability;
- video = i->capability & AST_FORMAT_VIDEO_MASK;
- text = i->capability & AST_FORMAT_TEXT_MASK;
+ if (!(ast_cap_is_empty(i->jointcaps))) { /* The joint capabilities of us and peer */
+ ast_cap_append(i->jointcaps, what);
+ video = ast_cap_get_type(i->jointcaps, AST_FORMAT_TYPE_VIDEO);
+ text = ast_cap_get_type(i->jointcaps, AST_FORMAT_TYPE_TEXT);
+ } else if (!(ast_cap_is_empty(i->caps))) { /* Our configured capability for this peer */
+ ast_cap_append(i->caps, what);
+ video = ast_cap_get_type(i->caps, AST_FORMAT_TYPE_VIDEO);
+ text = ast_cap_get_type(i->caps, AST_FORMAT_TYPE_TEXT);
} else {
- what = sip_cfg.capability; /* Global codec support */
- video = sip_cfg.capability & AST_FORMAT_VIDEO_MASK;
- text = sip_cfg.capability & AST_FORMAT_TEXT_MASK;
+ ast_cap_append(sip_cfg.caps, what); /* Global codec support */
+ video = ast_cap_get_type(sip_cfg.caps, AST_FORMAT_TYPE_VIDEO);
+ text = ast_cap_get_type(sip_cfg.caps, AST_FORMAT_TYPE_TEXT);
}
/* Set the native formats for audio and merge in video */
- tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text;
+ ast_codec_choose(&i->prefs, what, 1, &fmt);
+ ast_cap_append(video, tmp->nativeformats);
+ ast_cap_append(text, tmp->nativeformats);
+ ast_cap_add(tmp->nativeformats, &fmt);
ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats));
- ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability));
- ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));
- ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1)));
- if (i->prefcodec) {
- ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec));
+ ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcaps));
+ ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->caps));
+ ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname(&fmt));
+ if (!ast_cap_is_empty(i->prefcaps)) {
+ ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcaps));
}
/* XXX Why are we choosing a codec from the native formats?? */
- fmt = ast_best_codec(tmp->nativeformats);
+ ast_best_codec(tmp->nativeformats, &fmt);
/* If we have a prefcodec setting, we have an inbound channel that set a
preferred format for this call. Otherwise, we check the jointcapability
@@ -6607,18 +6624,18 @@
*/
if (i->vrtp) {
if (ast_test_flag(&i->flags[1], SIP_PAGE2_VIDEOSUPPORT))
- needvideo = AST_FORMAT_VIDEO_MASK;
- else if (i->prefcodec)
- needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK; /* Outbound call */
+ needvideo = 1;
+ else if (!ast_cap_is_empty(i->prefcaps))
+ needvideo = ast_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_VIDEO); /* Outbound call */
else
- needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */
+ needvideo = ast_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_VIDEO); /* Inbound call */
}
if (i->trtp) {
- if (i->prefcodec)
- needtext = i->prefcodec & AST_FORMAT_TEXT_MASK; /* Outbound call */
+ if (!ast_cap_is_empty(i->prefcaps))
+ needtext = ast_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_TEXT); /* Outbound call */
else
- needtext = i->jointcapability & AST_FORMAT_TEXT_MASK; /* Inbound call */
+ needtext = ast_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_TEXT); /* Inbound call */
}
if (needvideo) {
@@ -6661,13 +6678,13 @@
}
tmp->adsicpe = AST_ADSI_UNAVAILABLE;
- tmp->writeformat = fmt;
- tmp->rawwriteformat = fmt;
- ast_rtp_instance_set_write_format(i->rtp, fmt);
-
- tmp->readformat = fmt;
- tmp->rawreadformat = fmt;
- ast_rtp_instance_set_read_format(i->rtp, fmt);
+ ast_format_copy(&fmt, &tmp->writeformat);
+ ast_format_copy(&fmt, &tmp->rawwriteformat);
+ ast_rtp_instance_set_write_format(i->rtp, &fmt);
+
+ ast_format_copy(&fmt, &tmp->readformat);
+ ast_format_copy(&fmt, &tmp->rawreadformat);
+ ast_rtp_instance_set_read_format(i->rtp, &fmt);
tmp->tech_pvt = dialog_ref(i, "sip_new: set chan->tech_pvt to i");
@@ -6755,6 +6772,10 @@
"Channel: %s\r\nUniqueid: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\n",
tmp->name, tmp->uniqueid, "SIP", i->callid, i->fullcontact);
}
+
+ ast_cap_destroy(what);
+ ast_cap_destroy(video);
+ ast_cap_destroy(text);
return tmp;
}
@@ -6975,17 +6996,18 @@
return f;
}
- if (f && f->subclass.codec != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
- if (!(f->subclass.codec & p->jointcapability)) {
+ if (f && !ast_cap_iscompatible(p->owner->nativeformats, &f->subclass.format)) {
+ if (!ast_cap_iscompatible(p->jointcaps, &f->subclass.format)) {
ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
- ast_getformatname(f->subclass.codec), p->owner->name);
+ ast_getformatname(&f->subclass.format), p->owner->name);
return &ast_null_frame;
}
ast_debug(1, "Oooh, format changed to %s\n",
- ast_getformatname(f->subclass.codec));
- p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_MASK)) | f->subclass.codec;
- ast_set_read_format(p->owner, p->owner->readformat);
- ast_set_write_format(p->owner, p->owner->writeformat);
+ ast_getformatname(&f->subclass.format));
+ ast_cap_remove_bytype(p->owner->nativeformats, AST_FORMAT_TYPE_AUDIO);
+ ast_cap_add(p->owner->nativeformats, &f->subclass.format);
+ ast_set_read_format(p->owner, &p->owner->readformat);
+ ast_set_write_format(p->owner, &p->owner->writeformat);
}
if (f && p->dsp) {
@@ -7148,6 +7170,20 @@
ao2_t_ref(p, -1, "Yuck, couldn't allocate cc_params struct. Get rid o' p");
return NULL;
}
+ p->caps = ast_cap_alloc();
+ p->jointcaps = ast_cap_alloc();
+ p->peercaps = ast_cap_alloc();
+ p->redircaps = ast_cap_alloc();
+
+ if (!p->caps|| !p->jointcaps || !p->peercaps || !p->redircaps) {
+ p->caps = ast_cap_destroy(p->caps);
+ p->jointcaps = ast_cap_destroy(p->jointcaps);
+ p->peercaps = ast_cap_destroy(p->peercaps);
+ p->redircaps = ast_cap_destroy(p->redircaps);
+ ao2_t_ref(p, -1, "Yuck, couldn't allocate cc_params struct. Get rid o' p");
+ return NULL;
+ }
+
/* If this dialog is created as a result of a request or response, lets store
* some information about it in the dialog. */
@@ -7251,7 +7287,7 @@
/* Assign default music on hold class */
ast_string_field_set(p, mohinterpret, default_mohinterpret);
ast_string_field_set(p, mohsuggest, default_mohsuggest);
- p->capability = sip_cfg.capability;
+ ast_cap_append(sip_cfg.caps, p->caps);
p->allowtransfer = sip_cfg.allowtransfer;
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
@@ -8216,6 +8252,8 @@
*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action)
{
+ int res = 0;
+
/* Iterators for SDP parsing */
int start = req->sdp_start;
int next = start;
@@ -8238,18 +8276,21 @@
struct ast_sockaddr *vsa = NULL; /*!< RTP video host IP */
struct ast_sockaddr *tsa = NULL; /*!< RTP text host IP */
struct ast_sockaddr *isa = NULL; /*!< UDPTL host ip */
- int portno = -1; /*!< RTP Audio port number */
- int vportno = -1; /*!< RTP Video port number */
+ int portno = -1; /*!< RTP Audio port number */
+ int vportno = -1; /*!< RTP Video port number */
int tportno = -1; /*!< RTP Text port number */
int udptlportno = -1; /*!< UDPTL Image port number */
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
- format_t peercapability = 0, vpeercapability = 0, tpeercapability = 0;
+ struct ast_cap *peercapability = ast_cap_alloc();
+ struct ast_cap *vpeercapability = ast_cap_alloc();
+ struct ast_cap *tpeercapability = ast_cap_alloc();
+
int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;
struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp;
- format_t newjointcapability; /* Negotiated capability */
- format_t newpeercapability;
+ struct ast_cap *newjointcapability = ast_cap_alloc(); /* Negotiated capability */
+ struct ast_cap *newpeercapability = ast_cap_alloc();
int newnoncodeccapability;
const char *codecs;
@@ -8272,12 +8313,18 @@
/* START UNKNOWN */
char buf[SIPBUFSIZE];
+ struct ast_format tmp_fmt;
/* END UNKNOWN */
/* Initial check */
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
- return -1;
+ res = -1;
+ goto process_sdp_cleanup;
+ }
+ if (!peercapability || !vpeercapability || !tpeercapability || !newpeercapability || !newjointcapability) {
+ res = -1;
+ goto process_sdp_cleanup;
}
/* Make sure that the codec structures are all cleared out */
@@ -8307,7 +8354,8 @@
nextm = get_sdp_iterate(&next, req, "m");
if (ast_strlen_zero(nextm)) {
ast_log(LOG_WARNING, "Insufficient information for SDP (m= not found)\n");
- return -1;
+ res = -1;
+ goto process_sdp_cleanup;
}
/* Scan session level SDP parameters (lines before first media stream) */
@@ -8319,7 +8367,8 @@
* error. We just want to ignore the SDP and let the rest of the packet be handled as normal.
*/
if (!process_sdp_o(value, p))
- return (p->session_modify == FALSE) ? 0 : -1;
+ res = (p->session_modify == FALSE) ? 0 : -1;
+ goto process_sdp_cleanup;
break;
case 'c':
if (process_sdp_c(value, &sessionsa)) {
@@ -8387,7 +8436,8 @@
for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
- return -1;
+ res = -1;
+ goto process_sdp_cleanup;
}
if (debug)
ast_verbose("Found RTP audio format %d\n", codec);
@@ -8415,7 +8465,8 @@
for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
- return -1;
+ res = -1;
+ goto process_sdp_cleanup;
}
if (debug)
ast_verbose("Found RTP video format %d\n", codec);
@@ -8436,7 +8487,8 @@
for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
- return -1;
+ res = -1;
+ goto process_sdp_cleanup;
}
if (debug)
ast_verbose("Found RTP text format %d\n", codec);
@@ -8541,45 +8593,53 @@
/* Sanity checks */
if (!sa && !vsa && !tsa && !isa) {
ast_log(LOG_WARNING, "Insufficient information in SDP (c=)...\n");
- return -1;
+ res = -1;
+ goto process_sdp_cleanup;
}
if (portno == -1 && vportno == -1 && udptlportno == -1 && tportno == -1) {
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
ast_log(LOG_WARNING, "Failing due to no acceptable offer found\n");
- return -2;
+ res = -2;
+ goto process_sdp_cleanup;
}
if (numberofmediastreams > 3) {
/* We have too many fax, audio and/or video and/or text media streams, fail this offer */
ast_log(LOG_WARNING, "Faling due to too many media streams\n");
- return -3;
+ res = -3;
+ goto process_sdp_cleanup;
}
if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) {
ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
- return -4;
+ res = -4;
+ goto process_sdp_cleanup;
}
if (!secure_audio && p->srtp) {
ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
- return -4;
+ res = -4;
+ goto process_sdp_cleanup;
}
if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) {
ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
- return -4;
+ res = -4;
+ goto process_sdp_cleanup;
}
if (!p->novideo && !secure_video && p->vsrtp) {
ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
- return -4;
+ res = -4;
+ goto process_sdp_cleanup;
}
if (!(secure_audio || secure_video) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n");
- return -4;
+ res = -4;
+ goto process_sdp_cleanup;
}
if (udptlportno == -1) {
@@ -8587,12 +8647,22 @@
}
/* Now gather all of the codecs that we are asked for: */
- ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
- ast_rtp_codecs_payload_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
- ast_rtp_codecs_payload_formats(&newtextrtp, &tpeercapability, &tpeernoncodeccapability);
-
- newjointcapability = p->capability & (peercapability | vpeercapability | tpeercapability);
- newpeercapability = (peercapability | vpeercapability | tpeercapability);
+ ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability);
+ ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability);
+ ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability);
+
+ ast_cap_append(newpeercapability, peercapability);
+ ast_cap_append(newpeercapability, vpeercapability);
+ ast_cap_append(newpeercapability, tpeercapability);
+
+ ast_cap_joint_copy(p->caps, newpeercapability, newjointcapability);
+ if (ast_cap_is_empty(newjointcapability) && (portno != -1)) {
+ ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
+ /* Do NOT Change current setting */
+ res = -1;
+ goto process_sdp_cleanup;
+ }
+
newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
if (debug) {
@@ -8600,7 +8670,7 @@
char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE];
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
- ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability),
+ ast_getformatname_multiple(s1, SIPBUFSIZE, p->caps),
ast_getformatname_multiple(s2, SIPBUFSIZE, peercapability),
ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
@@ -8612,14 +8682,9 @@
struct ast_str *s3 = ast_str_alloca(SIPBUFSIZE);
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
- ast_rtp_lookup_mime_multiple2(s1, p->noncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple2(s2, peernoncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple2(s3, newnoncodeccapability, 0, 0));
- }
- if (!newjointcapability && (portno != -1)) {
- ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
- /* Do NOT Change current setting */
- return -1;
+ ast_rtp_lookup_mime_multiple2(s1, NULL, p->noncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple2(s2, NULL, peernoncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
}
/* Setup audio address and port */
@@ -8633,12 +8698,18 @@
}
/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
they are acceptable */
- p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
- p->peercapability = newpeercapability; /* The other sides capability in latest offer */
+ ast_cap_remove_all(p->jointcaps);
+ ast_cap_append(newjointcapability, p->jointcaps); /* Our joint codec profile for this call */
+
+ ast_cap_remove_all(p->peercaps);
+ ast_cap_append(newpeercapability, p->peercaps); /* The other sides capability in latest offer */
+
p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */
- p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
+ ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt);
+ ast_cap_remove_all(p->jointcaps);
+ ast_cap_add(p->jointcaps, &tmp_fmt);
}
ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
@@ -8691,7 +8762,7 @@
ast_verbose("Peer T.140 RTP is at port %s\n",
ast_sockaddr_stringify(tsa));
}
- if ((p->jointcapability & AST_FORMAT_T140RED)) {
+ if (ast_cap_iscompatible(p->jointcaps, ast_format_set(&tmp_fmt, AST_FORMAT_T140RED, 0))) {
p->red = 1;
ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
} else {
@@ -8761,27 +8832,37 @@
if ((portno == -1) && (p->t38.state != T38_DISABLED)) {
ast_debug(3, "Have T.38 but no audio, accepting offer anyway\n");
- return 0;
- }
+ res = 0;
+ goto process_sdp_cleanup;
+ }
/* Ok, we're going with this offer */
- ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));
-
- if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
- return 0;
+ ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcaps));
+
+ if (!p->owner) { /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
+ res = 0;
+ goto process_sdp_cleanup;
+ }
ast_debug(4, "We have an owner, now see if we need to change this call\n");
- if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
+ if (!(ast_cap_has_joint(p->owner->nativeformats, p->jointcaps)) && ast_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO)) {
if (debug) {
char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n",
- ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability),
+ ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcaps),
ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats));
}
- p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability) | (p->capability & tpeercapability);
- ast_set_read_format(p->owner, p->owner->readformat);
- ast_set_write_format(p->owner, p->owner->writeformat);
+
+ ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt);
+ ast_cap_remove_all(p->owner->nativeformats);
+
+ ast_cap_add(p->owner->nativeformats, &tmp_fmt);
+ ast_cap_joint_copy(p->caps, vpeercapability, p->owner->nativeformats);
+ ast_cap_joint_copy(p->caps, tpeercapability, p->owner->nativeformats);
+
+ ast_set_read_format(p->owner, &p->owner->readformat);
+ ast_set_write_format(p->owner, &p->owner->writeformat);
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && (!ast_sockaddr_isnull(sa) || !ast_sockaddr_isnull(vsa) || !ast_sockaddr_isnull(tsa) || !ast_sockaddr_isnull(isa)) && (!sendonly || sendonly == -1)) {
@@ -8800,8 +8881,14 @@
ast_queue_frame(p->owner, &ast_null_frame);
change_hold_state(p, req, TRUE, sendonly);
}
-
- return 0;
+
+process_sdp_cleanup:
+ ast_cap_destroy(peercapability);
+ ast_cap_destroy(vpeercapability);
+ ast_cap_destroy(tpeercapability);
+ ast_cap_destroy(newjointcapability);
+ ast_cap_destroy(newpeercapability);
+ return res;
}
static int process_sdp_o(const char *o, struct sip_pvt *p)
@@ -8955,10 +9042,10 @@
int codec_n;
for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
- if (!format.asterisk_format || !format.code) /* non-codec or not found */
+ if (!format.asterisk_format) /* non-codec or not found */
continue;
- ast_debug(1, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing);
- ast_codec_pref_setsize(pref, format.code, framing);
+ ast_debug(1, "Setting framing for %s to %ld\n", ast_getformatname(&format.format), framing);
+ ast_codec_pref_setsize(pref, &format.format, framing);
}
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
}
@@ -8986,10 +9073,10 @@
struct ast_rtp_payload_type payload;
payload = ast_rtp_codecs_payload_lookup(newaudiortp, codec);
- if (payload.code && payload.asterisk_format) {
+ if (payload.format.id && payload.asterisk_format) {
unsigned int bit_rate;
- switch (payload.code) {
+ switch ((int) payload.format.id) {
case AST_FORMAT_SIREN7:
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 32000) {
@@ -10318,31 +10405,34 @@
}
/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
- struct ast_str **m_buf, struct ast_str **a_buf,
- int debug, int *min_packet_size)
+static void add_codec_to_sdp(const struct sip_pvt *p,
+ struct ast_format *format,
+ struct ast_str **m_buf,
+ struct ast_str **a_buf,
+ int debug,
+ int *min_packet_size)
{
int rtp_code;
struct ast_format_list fmt;
if (debug)
- ast_verbose("Adding codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec));
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
+ ast_verbose("Adding codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, format, 0)) == -1)
return;
if (p->rtp) {
struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
- fmt = ast_codec_pref_getsize(pref, codec);
+ fmt = ast_codec_pref_getsize(pref, format);
} else /* I don't see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
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_subtype2(1, codec,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
- ast_rtp_lookup_sample_rate2(1, codec));
-
- switch (codec) {
+ ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n",
+ rtp_code,
+ ast_rtp_lookup_mime_subtype2(1, format, 0, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
+ ast_rtp_lookup_sample_rate2(1, format, 0));
+
+ switch ((int) format->id) {
case 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);
@@ -10379,7 +10469,7 @@
/*! \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, format_t codec,
+static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -10389,20 +10479,20 @@
return;
if (debug)
- ast_verbose("Adding video codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec));
-
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1)
+ ast_verbose("Adding video codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
+
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, format, 0)) == -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_subtype2(1, codec, 0),
- ast_rtp_lookup_sample_rate2(1, codec));
+ ast_rtp_lookup_mime_subtype2(1, format, 0, 0),
+ ast_rtp_lookup_sample_rate2(1, format, 0));
/* 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,
+static void add_tcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -10412,19 +10502,20 @@
return;
if (debug)
- ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
-
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, codec)) == -1)
+ ast_verbose("Adding text codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
+
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, format, 0)) == -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_subtype2(1, codec, 0),
- ast_rtp_lookup_sample_rate2(1, codec));
+ ast_rtp_lookup_mime_subtype2(1, format, 0, 0),
+ ast_rtp_lookup_sample_rate2(1, format, 0));
/* Add fmtp code here */
- if (codec == AST_FORMAT_T140RED) {
- int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, AST_FORMAT_T140);
+ if (format->id == AST_FORMAT_T140RED) {
+ struct ast_format tmp_fmt;
+ int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, ast_format_set(&tmp_fmt, AST_FORMAT_T140, 0), 0);
ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
t140code,
t140code,
@@ -10463,14 +10554,14 @@
int rtp_code;
if (debug)
- ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, format, 0));
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, format)) == -1)
+ ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, NULL, format, 0));
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, NULL, format)) == -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_subtype2(0, format, 0),
- ast_rtp_lookup_sample_rate2(0, format));
+ ast_rtp_lookup_mime_subtype2(0, NULL, format, 0),
+ ast_rtp_lookup_sample_rate2(0, NULL, format));
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);
}
@@ -10596,8 +10687,8 @@
*/
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
{
- format_t alreadysent = 0;
-
+ struct ast_cap *alreadysent = ast_cap_alloc();
+ int res = AST_SUCCESS;
struct ast_sockaddr addr = { {0,} };
struct ast_sockaddr vaddr = { {0,} };
struct ast_sockaddr taddr = { {0,} };
@@ -10627,8 +10718,9 @@
const char *v_a_crypto = NULL;
const char *t_a_crypto = NULL;
- format_t x;
- format_t capability = 0;
+ int x;
+ struct ast_format tmp_fmt;
+ struct ast_cap *capability = NULL; /* this holds a shallow copy and should never be destroyed */
int needaudio = FALSE;
int needvideo = FALSE;
int needtext = FALSE;
@@ -10644,9 +10736,16 @@
/* Set the SDP session name */
snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
+ if (!alreadysent) {
+ res = AST_FAILURE;
+ goto add_sdp_cleanup;
+ }
if (!p->rtp) {
+ ast_cap_destroy(alreadysent);
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
- return AST_FAILURE;
+ res = AST_FAILURE;
+ goto add_sdp_cleanup;
+
}
/* XXX We should not change properties in the SIP dialog until
we have acceptance of the offer if this is a re-invite */
@@ -10662,7 +10761,7 @@
if (add_audio) {
/* Check if we need video in this call */
- if ((p->jointcapability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
+ if ((ast_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_VIDEO)) && !p->novideo) {
if (p->vrtp) {
needvideo = TRUE;
ast_debug(2, "This call needs video offers!\n");
@@ -10670,7 +10769,7 @@
ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
}
/* Check if we need text in this call */
- if ((p->jointcapability & AST_FORMAT_TEXT_MASK) && !p->notext) {
[... 770 lines stripped ...]
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