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-rw-r--r--channels/chan_sip.c164
1 files changed, 81 insertions, 83 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8934c1a97..438df8f33 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1146,7 +1146,7 @@ static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code)
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
-#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
+#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
#endif
/*@}*/
@@ -1215,7 +1215,7 @@ struct sip_settings {
char default_context[AST_MAX_CONTEXT];
char default_subscribecontext[AST_MAX_CONTEXT];
struct ast_ha *contact_ha; /*! \brief Global list of addresses dynamic peers are not allowed to use */
- int capability; /*!< Supported codecs */
+ format_t capability; /*!< Supported codecs */
};
static struct sip_settings sip_cfg; /*!< SIP configuration data.
@@ -1767,13 +1767,13 @@ struct sip_pvt {
unsigned int sipoptions; /*!< Supported SIP options on the other end */
unsigned int reqsipoptions; /*!< Required SIP options on the other end */
struct ast_codec_pref prefs; /*!< codec prefs */
- int capability; /*!< Special capability (codec) */
- int jointcapability; /*!< Supported capability at both ends (codecs) */
- int peercapability; /*!< Supported peer capability */
- int prefcodec; /*!< Preferred codec (outbound only) */
+ format_t capability; /*!< Special capability (codec) */
+ format_t jointcapability; /*!< Supported capability at both ends (codecs) */
+ format_t peercapability; /*!< Supported peer capability */
+ format_t prefcodec; /*!< Preferred codec (outbound only) */
int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
int jointnoncodeccapability; /*!< Joint Non codec capability */
- int redircodecs; /*!< Redirect codecs */
+ format_t redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */
@@ -2038,7 +2038,7 @@ struct sip_peer {
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
int expire; /*!< When to expire this peer registration */
- int capability; /*!< Codec capability */
+ format_t capability; /*!< Codec capability */
int rtptimeout; /*!< RTP timeout */
int rtpholdtimeout; /*!< RTP Hold Timeout */
int rtpkeepalive; /*!< Send RTP packets for keepalive */
@@ -2408,7 +2408,7 @@ enum t38_action_flag {
in coming releases. */
/*--- PBX interface functions */
-static struct ast_channel *sip_request_call(const char *type, int format, const struct ast_channel *requestor, void *data, int *cause);
+static struct ast_channel *sip_request_call(const char *type, format_t format, 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);
@@ -2501,7 +2501,7 @@ static const char* get_sdp_iterate(int* start, struct sip_request *req, const ch
static const char *get_sdp(struct sip_request *req, const char *name);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
-static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
+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_noncodec_to_sdp(const struct sip_pvt *p, int format,
@@ -2769,7 +2769,7 @@ static enum st_mode st_get_mode(struct sip_pvt *);
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, int 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, format_t codecs, int nat_active);
/*!--- SIP MWI Subscription support */
static int sip_subscribe_mwi(const char *value, int lineno);
@@ -6459,7 +6459,7 @@ static int sip_hangup(struct ast_channel *ast)
/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
static void try_suggested_sip_codec(struct sip_pvt *p)
{
- int fmt;
+ format_t fmt;
const char *codec;
struct ast_channel* chan;
@@ -6526,16 +6526,13 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
switch (frame->frametype) {
case AST_FRAME_VOICE:
- if (!(frame->subclass & ast->nativeformats)) {
+ if (!(frame->subclass.codec & ast->nativeformats)) {
char s1[512], s2[512], s3[512];
- ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %s(%d) read/write = %s(%d)/%s(%d)\n",
- frame->subclass,
- ast_getformatname_multiple(s1, sizeof(s1) - 1, ast->nativeformats & AST_FORMAT_AUDIO_MASK),
- ast->nativeformats & AST_FORMAT_AUDIO_MASK,
- ast_getformatname_multiple(s2, sizeof(s2) - 1, ast->readformat),
- ast->readformat,
- ast_getformatname_multiple(s3, sizeof(s3) - 1, ast->writeformat),
- ast->writeformat);
+ 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));
return 0;
}
if (p) {
@@ -6917,11 +6914,11 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
{
struct ast_channel *tmp;
struct ast_variable *v = NULL;
- int fmt;
- int what;
- int video;
- int text;
- int needvideo = 0;
+ format_t fmt;
+ format_t what;
+ format_t video;
+ format_t text;
+ format_t needvideo = 0;
int needtext = 0;
char buf[SIPBUFSIZE];
char *decoded_exten;
@@ -7285,7 +7282,7 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
/* Don't forward RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF_BEGIN || f->frametype == AST_FRAME_DTMF_END) &&
(ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833)) {
- ast_debug(1, "Ignoring DTMF (%c) RTP frame because dtmfmode is not RFC2833\n", f->subclass);
+ ast_debug(1, "Ignoring DTMF (%c) RTP frame because dtmfmode is not RFC2833\n", f->subclass.integer);
return &ast_null_frame;
}
@@ -7293,15 +7290,15 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
if (!p->owner || (f && f->frametype != AST_FRAME_VOICE))
return f;
- if (f && f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
- if (!(f->subclass & p->jointcapability)) {
+ if (f && f->subclass.codec != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
+ if (!(f->subclass.codec & p->jointcapability)) {
ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
- ast_getformatname(f->subclass), p->owner->name);
+ ast_getformatname(f->subclass.codec), p->owner->name);
return &ast_null_frame;
}
- ast_debug(1, "Oooh, format changed to %d %s\n",
- f->subclass, ast_getformatname(f->subclass));
- p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_MASK)) | f->subclass;
+ 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);
}
@@ -7309,11 +7306,11 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
if (f && p->dsp) {
f = ast_dsp_process(p->owner, p->dsp, f);
if (f && f->frametype == AST_FRAME_DTMF) {
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && f->subclass == 'f') {
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && f->subclass.integer == 'f') {
ast_debug(1, "Fax CNG detected on %s\n", ast->name);
*faxdetect = 1;
} else {
- ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass);
+ ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass.integer);
}
}
}
@@ -8367,9 +8364,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int old = 0;
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
- int peercapability = 0, peernoncodeccapability = 0;
- int vpeercapability = 0, vpeernoncodeccapability = 0;
- int tpeercapability = 0, tpeernoncodeccapability = 0;
+ format_t peercapability = 0, vpeercapability = 0, tpeercapability = 0;
+ int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;
struct sockaddr_in sin; /*!< media socket address */
struct sockaddr_in vsin; /*!< Video socket address */
struct sockaddr_in tsin; /*!< Text socket address */
@@ -8387,8 +8383,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int sendonly = -1;
int numberofports;
struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp;
- int newjointcapability; /* Negotiated capability */
- int newpeercapability;
+ format_t newjointcapability; /* Negotiated capability */
+ format_t newpeercapability;
int newnoncodeccapability;
int numberofmediastreams = 0;
int debug = sip_debug_test_pvt(p);
@@ -8781,7 +8777,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (!format.asterisk_format || !format.code) /* non-codec or not found */
continue;
if (option_debug)
- ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format.code, framing);
+ ast_log(LOG_DEBUG, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing);
ast_codec_pref_setsize(pref, format.code, framing);
}
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
@@ -10189,7 +10185,7 @@ static int add_vidupdate(struct sip_request *req)
}
/*! \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, int codec,
+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)
{
@@ -10198,7 +10194,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
if (debug)
- ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+ ast_verbose("Adding codec 0x%Lx (%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)
return;
@@ -10396,7 +10392,7 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
{
int len = 0;
- int alreadysent = 0;
+ format_t alreadysent = 0;
struct sockaddr_in sin = { 0, };
struct sockaddr_in vsin = { 0, };
@@ -10424,8 +10420,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */
struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
- int x;
- int capability = 0;
+ format_t x;
+ format_t capability = 0;
int needaudio = FALSE;
int needvideo = FALSE;
int needtext = FALSE;
@@ -10547,15 +10543,15 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
Note that p->prefcodec can include video codecs, so mask them out
*/
if (capability & p->prefcodec) {
- int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
+ format_t codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
/* Start by sending our preferred audio/video codecs */
- for (x = 0; x < 32; x++) {
- int codec;
+ for (x = 0; x < 64; x++) {
+ format_t codec;
if (!(codec = ast_codec_pref_index(&p->prefs, x)))
break;
@@ -10571,7 +10567,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
}
/* Now send any other common audio and video codecs, and non-codec formats: */
- for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
+ for (x = 1LL; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
if (!(capability & x)) /* Codec not requested */
continue;
@@ -10587,7 +10583,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
}
/* Now add DTMF RFC2833 telephony-event as a codec */
- for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
+ for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) {
if (!(p->jointnoncodeccapability & x))
continue;
@@ -14807,7 +14803,7 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req)
ast_verbose("SIP Text message received: '%s'\n", buf);
memset(&f, 0, sizeof(f));
f.frametype = AST_FRAME_TEXT;
- f.subclass = 0;
+ f.subclass.integer = 0;
f.offset = 0;
f.data.ptr = buf;
f.datalen = strlen(buf);
@@ -15669,9 +15665,10 @@ static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli
/*! \brief Print codec list from preference to CLI/manager */
static void print_codec_to_cli(int fd, struct ast_codec_pref *pref)
{
- int x, codec;
+ int x;
+ format_t codec;
- for(x = 0; x < 32 ; x++) {
+ for(x = 0; x < 64 ; x++) {
codec = ast_codec_pref_index(pref, x);
if (!codec)
break;
@@ -15864,7 +15861,8 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
struct ast_codec_pref *pref;
struct ast_variable *v;
struct sip_auth *auth;
- int x = 0, codec = 0, load_realtime;
+ int x = 0, load_realtime;
+ format_t codec = 0;
int realtimepeers;
realtimepeers = ast_check_realtime("sippeers");
@@ -16071,12 +16069,12 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
astman_append(s, "%s\r\n", codec_buf);
astman_append(s, "CodecOrder: ");
pref = &peer->prefs;
- for(x = 0; x < 32 ; x++) {
+ for(x = 0; x < 64 ; x++) {
codec = ast_codec_pref_index(pref, x);
if (!codec)
break;
astman_append(s, "%s", ast_getformatname(codec));
- if (x < 31 && ast_codec_pref_index(pref, x+1))
+ if (x < 63 && ast_codec_pref_index(pref, x+1))
astman_append(s, ",");
}
@@ -16965,10 +16963,10 @@ static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_a
ast_cli(a->fd, " Curr. trans. direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming");
ast_cli(a->fd, " Call-ID: %s\n", cur->callid);
ast_cli(a->fd, " Owner channel ID: %s\n", cur->owner ? cur->owner->name : "<none>");
- ast_cli(a->fd, " Our Codec Capability: %d\n", cur->capability);
+ ast_cli(a->fd, " Our Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->capability));
ast_cli(a->fd, " Non-Codec Capability (DTMF): %d\n", cur->noncodeccapability);
- ast_cli(a->fd, " Their Codec Capability: %d\n", cur->peercapability);
- ast_cli(a->fd, " Joint Codec Capability: %d\n", cur->jointcapability);
+ ast_cli(a->fd, " Their Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->peercapability));
+ ast_cli(a->fd, " Joint Codec Capability: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->jointcapability));
ast_cli(a->fd, " Format: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) );
ast_cli(a->fd, " T.38 support %s\n", cli_yesno(cur->udptl != NULL));
ast_cli(a->fd, " Video support %s\n", cli_yesno(cur->vrtp != NULL));
@@ -17178,7 +17176,7 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
event = atoi(buf);
if (event == 16) {
/* send a FLASH event */
- struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, };
+ struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
ast_queue_frame(p->owner, &f);
if (sipdebug)
ast_verbose("* DTMF-relay event received: FLASH\n");
@@ -17186,18 +17184,18 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
/* send a DTMF event */
struct ast_frame f = { AST_FRAME_DTMF, };
if (event < 10) {
- f.subclass = '0' + event;
- } else if (event < 11) {
- f.subclass = '*';
- } else if (event < 12) {
- f.subclass = '#';
+ f.subclass.integer = '0' + event;
+ } else if (event == 10) {
+ f.subclass.integer = '*';
+ } else if (event == 11) {
+ f.subclass.integer = '#';
} else if (event < 16) {
- f.subclass = 'A' + (event - 12);
+ f.subclass.integer = 'A' + (event - 12);
}
f.len = duration;
ast_queue_frame(p->owner, &f);
if (sipdebug)
- ast_verbose("* DTMF-relay event received: %c\n", f.subclass);
+ ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
}
transmit_response(p, "200 OK", req);
return;
@@ -17221,7 +17219,7 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
event = atoi(buf);
if (event == 16) {
/* send a FLASH event */
- struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, };
+ struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH }, };
ast_queue_frame(p->owner, &f);
if (sipdebug)
ast_verbose("* DTMF-relay event received: FLASH\n");
@@ -17229,18 +17227,18 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
/* send a DTMF event */
struct ast_frame f = { AST_FRAME_DTMF, };
if (event < 10) {
- f.subclass = '0' + event;
- } else if (event < 11) {
- f.subclass = '*';
- } else if (event < 12) {
- f.subclass = '#';
+ f.subclass.integer = '0' + event;
+ } else if (event == 10) {
+ f.subclass.integer = '*';
+ } else if (event == 11) {
+ f.subclass.integer = '#';
} else if (event < 16) {
- f.subclass = 'A' + (event - 12);
+ f.subclass.integer = 'A' + (event - 12);
}
f.len = duration;
ast_queue_frame(p->owner, &f);
if (sipdebug)
- ast_verbose("* DTMF-relay event received: %c\n", f.subclass);
+ ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
}
transmit_response(p, "200 OK", req);
return;
@@ -17287,10 +17285,10 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
/* Send the feature code to the PBX as DTMF, just like the handset had sent it */
f.len = 100;
for (j=0; j < strlen(feat->exten); j++) {
- f.subclass = feat->exten[j];
+ f.subclass.integer = feat->exten[j];
ast_queue_frame(p->owner, &f);
if (sipdebug)
- ast_verbose("* DTMF-relay event faked: %c\n", f.subclass);
+ ast_verbose("* DTMF-relay event faked: %c\n", f.subclass.integer);
}
ast_unlock_call_features();
@@ -17866,7 +17864,7 @@ static int function_sippeer(struct ast_channel *chan, const char *cmd, char *dat
}
} else if (!strncasecmp(colname, "codec[", 6)) {
char *codecnum;
- int codec = 0;
+ format_t codec = 0;
codecnum = colname + 6; /* move past the '[' */
codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */
@@ -23955,7 +23953,7 @@ static int sip_devicestate(void *data)
* or SIP/host!dnid
* \endverbatim
*/
-static struct ast_channel *sip_request_call(const char *type, int format, const struct ast_channel *requestor, void *data, int *cause)
+static struct ast_channel *sip_request_call(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
{
struct sip_pvt *p;
struct ast_channel *tmpc = NULL;
@@ -23968,7 +23966,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, const
char *authname = NULL;
char *trans = NULL;
enum sip_transport transport = 0;
- int oldformat = format;
+ format_t oldformat = format;
/* mask request with some set of allowed formats.
* XXX this needs to be fixed.
@@ -26251,7 +26249,7 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru
return res;
}
-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, int 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, format_t codecs, int nat_active)
{
struct sip_pvt *p;
int changed = 0;
@@ -26321,7 +26319,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
return 0;
}
-static int sip_get_codec(struct ast_channel *chan)
+static format_t sip_get_codec(struct ast_channel *chan)
{
struct sip_pvt *p = chan->tech_pvt;
return p->peercapability ? p->peercapability : p->capability;
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-14 12:14:46 +0000