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Merged revisions 210190 via svnmerge from

Browse Source 
https://origsvn.digium.com/svn/asterisk/trunk

........
  r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines
  
  Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
  
  It is clear from multiple mailing list, forum, wiki and other sorts of posts
  that users don't really understand the effects that the 'canreinvite' config
  option actually has, and that in some cases they think that setting it to 'no'
  will actually cause various other features (T.38, MOH, etc.) to not work properly,
  when in fact this is not the case. This patch changes the proper name of the
  option to what it should have been from the beginning ('directmedia'), but
  preserves backwards compatibility for existing configurations.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.2
Kevin P. Fleming 17 years ago
parent f354439f99
commit bbf4a08fe7
13 changed files with 75 additions and 63 deletions
Show Stats Download Patch File Download Diff File

11
UPGRADE.txt
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@ -30,6 +30,17 @@ From 1.6.1 to 1.6.2:
Asterisk and various SIP/T.38 endpoints identified by many users of
Asterisk.
* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
has been renamed to 'directmedia', to better reflect what it actually does.
In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
option never had any effect on these cases, it only affected the re-INVITEs
used for direct media path setup. For MGCP and Skinny, the option was poorly
named because those protocols don't even use INVITE messages at all. For
backwards compatibility, the old option is still supported in both normal
and Realtime configuration files, but all of the sample configuration files,
Realtime/LDAP schemas, and other documentation refer to it using the new name.
* The default console now will use colors according to the default background
color, instead of forcing the background color to black. If you are using a
light colored background for your console, you may wish to use the option

18
channels/chan_mgcp.c
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@ -80,7 +80,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#define MGCPDUMPER
#define DEFAULT_EXPIRY 120
#define MAX_EXPIRY 3600
#define CANREINVITE 1
#define DIRECTMEDIA 1
#ifndef INADDR_NONE
#define INADDR_NONE (in_addr_t)(-1)
@ -177,7 +177,7 @@ static int cancallforward = 0;
static int singlepath = 0;
static int canreinvite = CANREINVITE;
static int directmedia = DIRECTMEDIA;
static char accountcode[AST_MAX_ACCOUNT_CODE] = "";
@ -330,7 +330,7 @@ struct mgcp_endpoint {
int threewaycalling;
int singlepath;
int cancallforward;
int canreinvite;
int directmedia;
int callreturn;
int dnd; /* How does this affect callwait? Do we just deny a mgcp_request if we're dnd? */
int hascallerid;
@ -3557,7 +3557,7 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
int i=0, y=0;
int gw_reload = 0;
int ep_reload = 0;
canreinvite = CANREINVITE;
directmedia = DIRECTMEDIA;
/* locate existing gateway */
gw = gateways;
@ -3667,8 +3667,8 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
cancallforward = ast_true(v->value);
} else if (!strcasecmp(v->name, "singlepath")) {
singlepath = ast_true(v->value);
} else if (!strcasecmp(v->name, "canreinvite")) {
canreinvite = ast_true(v->value);
} else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
directmedia = ast_true(v->value);
} else if (!strcasecmp(v->name, "mailbox")) {
ast_copy_string(mailbox, v->value, sizeof(mailbox));
} else if (!strcasecmp(v->name, "hasvoicemail")) {
@ -3753,7 +3753,7 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
e->callreturn = callreturn;
e->cancallforward = cancallforward;
e->singlepath = singlepath;
e->canreinvite = canreinvite;
e->directmedia = directmedia;
e->callwaiting = callwaiting;
e->hascallwaiting = callwaiting;
e->slowsequence = slowsequence;
@ -3856,7 +3856,7 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
e->pickupgroup=cur_pickupgroup;
e->callreturn = callreturn;
e->cancallforward = cancallforward;
e->canreinvite = canreinvite;
e->directmedia = directmedia;
e->singlepath = singlepath;
e->callwaiting = callwaiting;
e->hascallwaiting = callwaiting;
@ -3949,7 +3949,7 @@ static enum ast_rtp_get_result mgcp_get_rtp_peer(struct ast_channel *chan, struc
*rtp = sub->rtp;
if (sub->parent->canreinvite)
if (sub->parent->directmedia)
return AST_RTP_TRY_NATIVE;
else
return AST_RTP_TRY_PARTIAL;

41
channels/chan_sip.c
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@ -1335,8 +1335,8 @@ struct sip_auth {
/* re-INVITE related settings */
#define SIP_REINVITE (7 << 20) /*!< DP: four settings, uses three bits */
#define SIP_REINVITE_NONE (0 << 20) /*!< DP: no reinvite allowed */
#define SIP_CAN_REINVITE (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
#define SIP_CAN_REINVITE_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
#define SIP_DIRECT_MEDIA (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
#define SIP_DIRECT_MEDIA_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
#define SIP_REINVITE_UPDATE (4 << 20) /*!< DP: use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings - see insecure2str() */
@ -14796,7 +14796,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
ast_cli(fd, " ACL : %s\n", cli_yesno(peer->ha != NULL));
ast_cli(fd, " T38 pt UDPTL : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
ast_cli(fd, " CanReinvite : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)));
ast_cli(fd, " DirectMedia : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
ast_cli(fd, " PromiscRedir : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
ast_cli(fd, " User=Phone : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
ast_cli(fd, " Video Support: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)));
@ -14899,7 +14899,8 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
astman_append(s, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N"));
astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Y":"N"));
astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
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"));
@ -22535,11 +22536,11 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
else
ast_set_flag(&flags[0], SIP_NAT_RFC3581);
} else if (!strcasecmp(v->name, "canreinvite")) {
} else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
if (ast_true(v->value)) {
ast_set_flag(&flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT);
ast_set_flag(&flags[0], SIP_DIRECT_MEDIA | SIP_DIRECT_MEDIA_NAT);
} else if (!ast_false(v->value)) {
char buf[64];
char *word, *next = buf;
@ -22547,12 +22548,12 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
ast_copy_string(buf, v->value, sizeof(buf));
while ((word = strsep(&next, ","))) {
if (!strcasecmp(word, "update")) {
ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE);
ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_DIRECT_MEDIA);
} else if (!strcasecmp(word, "nonat")) {
ast_set_flag(&flags[0], SIP_CAN_REINVITE);
ast_clear_flag(&flags[0], SIP_CAN_REINVITE_NAT);
ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT);
} else {
ast_log(LOG_WARNING, "Unknown canreinvite mode '%s' on line %d\n", v->value, v->lineno);
ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno);
}
}
}
@ -23625,7 +23626,7 @@ static int reload_config(enum channelreloadreason reason)
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */
ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
/* Debugging settings, always default to off */
dumphistory = FALSE;
@ -24331,7 +24332,7 @@ static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
return NULL;
sip_pvt_lock(p);
if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
if (p->udptl && ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
udptl = p->udptl;
sip_pvt_unlock(p);
return udptl;
@ -24381,9 +24382,9 @@ static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct
*rtp = p->rtp;
if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT))
if (ast_rtp_getnat(*rtp) && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT))
res = AST_RTP_TRY_PARTIAL;
else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
res = AST_RTP_TRY_NATIVE;
else if (ast_test_flag(&global_jbconf, AST_JB_FORCED))
res = AST_RTP_GET_FAILED;
@ -24410,7 +24411,7 @@ static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struc
*rtp = p->vrtp;
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
res = AST_RTP_TRY_NATIVE;
sip_pvt_unlock(p);
@ -24435,7 +24436,7 @@ static enum ast_rtp_get_result sip_get_trtp_peer(struct ast_channel *chan, struc
*rtp = p->trtp;
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA))
res = AST_RTP_TRY_NATIVE;
sip_pvt_unlock(p);
@ -24467,10 +24468,10 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
/* if this peer cannot handle reinvites of the media stream to devices
that are known to be behind a NAT, then stop the process now
*/
if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
sip_pvt_unlock(p);
return 0;
}
if (nat_active && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
sip_pvt_unlock(p);
return 0;
}
if (rtp) {
changed |= ast_rtp_get_peer(rtp, &p->redirip);

12
channels/chan_skinny.c
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@ -1182,7 +1182,7 @@ struct skinny_subchannel {
int immediate; \
int hookstate; \
int nat; \
int canreinvite; \
int directmedia; \
int prune;
struct skinny_line {
@ -1208,7 +1208,7 @@ struct skinny_line_options{
.hidecallerid = 0,
.amaflags = 0,
.instance = 0,
.canreinvite = 0,
.directmedia = 0,
.nat = 0,
.confcapability = AST_FORMAT_ULAW | AST_FORMAT_ALAW,
.capability = 0,
@ -2593,7 +2593,7 @@ static enum ast_rtp_get_result skinny_get_rtp_peer(struct ast_channel *c, struct
l = sub->parent;
if (!l->canreinvite || l->nat){
if (!l->directmedia || l->nat){
res = AST_RTP_TRY_PARTIAL;
if (skinnydebug)
ast_verb(1, "skinny_get_rtp_peer() Using AST_RTP_TRY_PARTIAL \n");
@ -2653,7 +2653,7 @@ static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struc
req->data.startmedia.conferenceId = htolel(sub->callid);
req->data.startmedia.passThruPartyId = htolel(sub->callid);
if (!(l->canreinvite) || (l->nat)){
if (!(l->directmedia) || (l->nat)){
ast_rtp_get_us(rtp, &us);
req->data.startmedia.remoteIp = htolel(d->ourip.s_addr);
req->data.startmedia.remotePort = htolel(ntohs(us.sin_port));
@ -6631,9 +6631,9 @@ static struct ast_channel *skinny_request(const char *type, int format, void *da
CLINE_OPTS->callwaiting = ast_true(v->value);
continue;
}
} else if (!strcasecmp(v->name, "canreinvite")) {
} else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
if (type & (TYPE_DEF_LINE | TYPE_LINE)) {
CLINE_OPTS->canreinvite = ast_true(v->value);
CLINE_OPTS->directmedia = ast_true(v->value);
continue;
}
} else if (!strcasecmp(v->name, "nat")) {

6
configs/mgcp.conf.sample
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@ -41,7 +41,7 @@
;[dlinkgw]
;host = 192.168.0.64
;context = default
;canreinvite = no
;directmedia = no
;line => aaln/2
;line => aaln/1
@ -96,7 +96,7 @@
;callwaiting = no
;callreturn = yes
;cancallforward = yes
;canreinvite = no
;directmedia = no
;transfer = no
;dtmfmode = inband
;line => aaln/1 ; now lets save this config to line1 aka aaln/1
@ -104,7 +104,7 @@
;callwaiting = no
;callreturn = yes
;cancallforward = yes
;canreinvite = no
;directmedia = no
;transfer = no
;dtmfmode = inband
;line => aaln/2 ; now lets save this config to line2 aka aaln/2

4
configs/res_ldap.conf.sample
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@ -60,7 +60,7 @@ name = cn
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
canreinvite = AstAccountCanReinvite
directmedia = AstAccountDirectMedia
context = AstAccountContext
dtmfmode = AstAccountDTMFMode
fromuser = AstAccountFromUser
@ -131,7 +131,7 @@ additionalFilter=(objectClass=*)
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
canreinvite = AstAccountCanReinvite
directmedia = AstAccountDirectMedia
context = AstAccountContext
dtmfmode = AstAccountDTMFMode
fromuser = AstAccountFromUser

32
configs/sip.conf.sample
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@ -611,17 +611,17 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; ; (work around more UNIDEN bugs)
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work with in the case where Asterisk is outside and have
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
;
;canreinvite=yes ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason wants Asterisk to
; behind a NAT, or for some other reason want Asterisk to
; stay in the audio path, you may want to turn this off.
; This setting also affect direct RTP
@ -633,18 +633,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers INVITE. This will also fail if canreinvite is enabled when
; callers INVITE. This will also fail if directmedia is enabled when
; the device is actually behind NAT.
;canreinvite=nonat ; An additional option is to allow media path redirection
;directmedia=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).
;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
; instead of INVITE. This can be combined with 'nonat', as
; 'canreinvite=update,nonat'. It implies 'yes'.
; 'directmedia=update,nonat'. It implies 'yes'.
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
; number in SDP packets and will only modify the SDP
@ -808,7 +808,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; remotesecret
; transport
; dtmfmode
; canreinvite
; directmedia
; nat
; callgroup
; pickupgroup
@ -918,12 +918,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
[natted-phone](!,basic-options) ; another template inheriting basic-options
nat=yes
canreinvite=no
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
nat=no
canreinvite=yes
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
@ -958,7 +958,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
@ -988,7 +988,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;directmedia=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
@ -1061,7 +1061,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is

2
configs/skinny.conf.sample
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@ -157,7 +157,7 @@ keepalive=120
;device=SEP00D0BA847E6B
;version=P002G204 ; Thanks critch
;context=did
;canreinvite=yes ; Allow media to go directly between two RTP endpoints.
;directmedia=yes ; Allow media to go directly between two RTP endpoints.
;line=120 ; Dial(Skinny/120@florian)
; Typical config for a 7910

2
contrib/scripts/realtime_pgsql.sql
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@ -37,7 +37,7 @@ accountcode character varying(20),
amaflags character varying(7),
callgroup character varying(10),
callerid character varying(80),
canreinvite character varying(3) DEFAULT 'yes',
directmedia character varying(3) DEFAULT 'yes',
context character varying(80),
defaultip character varying(15),
dtmfmode character varying(7),

2
doc/chan_sip-perf-testing.txt
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@ -58,7 +58,7 @@ type=friend
context=test11
host=192.168.134.240 ;; the address of the host you will be running sipp on
user=sipp
canreinvite=no
directmedia=no
disallow=all
allow=ulaw

2
doc/res_config_sqlite.txt
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@ -70,7 +70,7 @@ CREATE TABLE ast_sip (
callgroup VARCHAR(10) DEFAULT NULL,
callerid VARCHAR(80) DEFAULT NULL,
cancallforward CHAR(3) DEFAULT 'yes',
canreinvite CHAR(3) DEFAULT 'yes',
directmedia CHAR(3) DEFAULT 'yes',
context VARCHAR(80) DEFAULT NULL,
defaultip VARCHAR(15) DEFAULT NULL,
dtmfmode VARCHAR(7) DEFAULT NULL,

2
doc/tex/phoneprov.tex
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@ -146,7 +146,7 @@ threewaycalling = yes
deletevoicemail = no
autoprov = yes
profile = polycom
canreinvite = no
directmedia = no
nat = no
fullname = User Two ; ${DISPLAY_NAME}
secret = test ; ${SECRET}

4
main/rtp.c
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@ -2124,7 +2124,7 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
text_src_res = srcpr->get_trtp_info ? srcpr->get_trtp_info(c1, &tsrcp) : AST_RTP_GET_FAILED;
}
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
/* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE)) {
/* Somebody doesn't want to play... */
ast_channel_unlock(c0);
@ -2211,7 +2211,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
else
destcodec = 0;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
/* Check if bridge is still possible (In SIP directmedia=no stops this, like NAT) */
if (audio_dest_res != AST_RTP_TRY_NATIVE || (video_dest_res != AST_RTP_GET_FAILED && video_dest_res != AST_RTP_TRY_NATIVE) || audio_src_res != AST_RTP_TRY_NATIVE || (video_src_res != AST_RTP_GET_FAILED && video_src_res != AST_RTP_TRY_NATIVE) || !(srccodec & destcodec)) {
/* Somebody doesn't want to play... */
ast_channel_unlock(dest);

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