Main Page | Alphabetical List | Data Structures | Directories | File List | Data Fields | Globals
include » asterisk

rtp.h File Reference

#include <asterisk/frame.h>
#include <asterisk/io.h>
#include <asterisk/sched.h>
#include <asterisk/channel.h>
#include <netinet/in.h>

Go to the source code of this file.

Data Structures

struct  ast_rtp_protocol

Defines

#define AST_RTP_DTMF   (1 << 0)
#define AST_RTP_CN   (1 << 1)
#define AST_RTP_CISCO_DTMF   (1 << 2)
#define AST_RTP_MAX   AST_RTP_CISCO_DTMF

Typedefs

typedef int(* ast_rtp_callback )(struct ast_rtp *rtp, struct ast_frame *f, void *data)

Functions

ast_rtp * ast_rtp_new (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
ast_rtp * ast_rtp_new_with_bindaddr (struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in)
void ast_rtp_set_peer (struct ast_rtp *rtp, struct sockaddr_in *them)
void ast_rtp_get_peer (struct ast_rtp *rtp, struct sockaddr_in *them)
void ast_rtp_get_us (struct ast_rtp *rtp, struct sockaddr_in *us)
void ast_rtp_destroy (struct ast_rtp *rtp)
void ast_rtp_set_callback (struct ast_rtp *rtp, ast_rtp_callback callback)
void ast_rtp_set_data (struct ast_rtp *rtp, void *data)
int ast_rtp_write (struct ast_rtp *rtp, struct ast_frame *f)
ast_frame * ast_rtp_read (struct ast_rtp *rtp)
ast_frame * ast_rtcp_read (struct ast_rtp *rtp)
int ast_rtp_fd (struct ast_rtp *rtp)
int ast_rtcp_fd (struct ast_rtp *rtp)
int ast_rtp_senddigit (struct ast_rtp *rtp, char digit)
int ast_rtp_settos (struct ast_rtp *rtp, int tos)
void ast_rtp_pt_clear (struct ast_rtp *rtp)
void ast_rtp_pt_default (struct ast_rtp *rtp)
void ast_rtp_set_m_type (struct ast_rtp *rtp, int pt)
void ast_rtp_set_rtpmap_type (struct ast_rtp *rtp, int pt, char *mimeType, char *mimeSubtype)
rtpPayloadType ast_rtp_lookup_pt (struct ast_rtp *rtp, int pt)
int ast_rtp_lookup_code (struct ast_rtp *rtp, int isAstFormat, int code)
void ast_rtp_offered_from_local (struct ast_rtp *rtp, int local)
void ast_rtp_get_current_formats (struct ast_rtp *rtp, int *astFormats, int *nonAstFormats)
char * ast_rtp_lookup_mime_subtype (int isAstFormat, int code)
void ast_rtp_setnat (struct ast_rtp *rtp, int nat)
int ast_rtp_bridge (struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
int ast_rtp_proto_register (struct ast_rtp_protocol *proto)
void ast_rtp_proto_unregister (struct ast_rtp_protocol *proto)
void ast_rtp_stop (struct ast_rtp *rtp)
void ast_rtp_init (void)
void ast_rtp_reload (void)

Define Documentation

#define AST_RTP_CISCO_DTMF   (1 << 2)
 

DTMF (Cisco Proprietary)

Definition at line 34 of file rtp.h.

Referenced by ast_rtp_read().

#define AST_RTP_CN   (1 << 1)
 

'Comfort Noise' (RFC3389)

Definition at line 32 of file rtp.h.

Referenced by ast_rtp_read().

#define AST_RTP_DTMF   (1 << 0)
 

DTMF (RFC2833)

Definition at line 30 of file rtp.h.

Referenced by ast_rtp_read(), and ast_rtp_senddigit().

#define AST_RTP_MAX   AST_RTP_CISCO_DTMF
 

Maximum RTP-specific code

Definition at line 36 of file rtp.h.

Typedef Documentation

typedef int(* ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data)
 

Definition at line 49 of file rtp.h.

Function Documentation

int ast_rtcp_fd (  struct ast_rtp *  rtp  ) 
 

Definition at line 119 of file rtp.c.

References ast_rtp::rtcp, and ast_rtcp::s.

00120 {
00121    if (rtp->rtcp)
00122       return rtp->rtcp->s;
00123    return -1;
00124 }

struct ast_frame* ast_rtcp_read (  struct ast_rtp *  rtp  ) 
 

Definition at line 335 of file rtp.c.

References AST_FRAME_NULL, ast_inet_ntoa(), ast_log(), CRASH, LOG_DEBUG, LOG_WARNING, ast_rtp::nat, option_debug, ast_rtp::rtcp, ast_rtcp::s, ast_rtp::them, and ast_rtcp::them.

00336 {
00337    static struct ast_frame null_frame = { AST_FRAME_NULL, };
00338    int len;
00339    int hdrlen = 8;
00340    int res;
00341    struct sockaddr_in sin;
00342    unsigned int rtcpdata[1024];
00343    char iabuf[INET_ADDRSTRLEN];
00344    
00345    if (!rtp->rtcp)
00346       return &null_frame;
00347 
00348    len = sizeof(sin);
00349    
00350    res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata),
00351                0, (struct sockaddr *)&sin, &len);
00352    
00353    if (res < 0) {
00354       if (errno != EAGAIN)
00355          ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
00356       if (errno == EBADF)
00357          CRASH;
00358       return &null_frame;
00359    }
00360 
00361    if (res < hdrlen) {
00362       ast_log(LOG_WARNING, "RTP Read too short\n");
00363       return &null_frame;
00364    }
00365 
00366    if (rtp->nat) {
00367       /* Send to whoever sent to us */
00368       if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
00369           (rtp->rtcp->them.sin_port != sin.sin_port)) {
00370          memcpy(&rtp->them, &sin, sizeof(rtp->them));
00371          ast_log(LOG_DEBUG, "RTP NAT: Using address %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
00372       }
00373    }
00374    if (option_debug)
00375       ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
00376    return &null_frame;
00377 }

int ast_rtp_bridge (  struct ast_channel *  c0,
struct ast_channel *  c1,
int  flags,
struct ast_frame **  fo,
struct ast_channel **  rc
) 
 

Definition at line 1351 of file rtp.c.

References ast_channel::_softhangup, AST_BRIDGE_DTMF_CHANNEL_0, AST_BRIDGE_DTMF_CHANNEL_1, ast_check_hangup(), AST_FRAME_DTMF, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frfree(), ast_inet_ntoa(), ast_log(), ast_mutex_lock, ast_mutex_trylock, ast_mutex_unlock, ast_read(), ast_rtp_get_peer(), ast_waitfor_n(), ast_write(), ast_frame::frametype, ast_rtp_protocol::get_codec, ast_rtp_protocol::get_rtp_info, ast_rtp_protocol::get_vrtp_info, ast_channel::lock, LOG_DEBUG, LOG_WARNING, ast_channel::masq, ast_channel::masqr, ast_channel::name, ast_channel_pvt::pvt, ast_channel::pvt, and ast_rtp_protocol::set_rtp_peer.

01352 {
01353    struct ast_frame *f;
01354    struct ast_channel *who, *cs[3];
01355    struct ast_rtp *p0, *p1;
01356    struct ast_rtp *vp0, *vp1;
01357    struct ast_rtp_protocol *pr0, *pr1;
01358    struct sockaddr_in ac0, ac1;
01359    struct sockaddr_in vac0, vac1;
01360    struct sockaddr_in t0, t1;
01361    struct sockaddr_in vt0, vt1;
01362    char iabuf[INET_ADDRSTRLEN];
01363    
01364    void *pvt0, *pvt1;
01365    int to;
01366    int codec0,codec1, oldcodec0, oldcodec1;
01367    
01368    memset(&vt0, 0, sizeof(vt0));
01369    memset(&vt1, 0, sizeof(vt1));
01370    memset(&vac0, 0, sizeof(vac0));
01371    memset(&vac1, 0, sizeof(vac1));
01372 
01373    /* if need DTMF, cant native bridge */
01374    if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
01375       return -2;
01376    ast_mutex_lock(&c0->lock);
01377    while(ast_mutex_trylock(&c1->lock)) {
01378       ast_mutex_unlock(&c0->lock);
01379       usleep(1);
01380       ast_mutex_lock(&c0->lock);
01381    }
01382    pr0 = get_proto(c0);
01383    pr1 = get_proto(c1);
01384    if (!pr0) {
01385       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
01386       ast_mutex_unlock(&c0->lock);
01387       ast_mutex_unlock(&c1->lock);
01388       return -1;
01389    }
01390    if (!pr1) {
01391       ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
01392       ast_mutex_unlock(&c0->lock);
01393       ast_mutex_unlock(&c1->lock);
01394       return -1;
01395    }
01396    pvt0 = c0->pvt->pvt;
01397    pvt1 = c1->pvt->pvt;
01398    p0 = pr0->get_rtp_info(c0);
01399    if (pr0->get_vrtp_info)
01400       vp0 = pr0->get_vrtp_info(c0);
01401    else
01402       vp0 = NULL;
01403    p1 = pr1->get_rtp_info(c1);
01404    if (pr1->get_vrtp_info)
01405       vp1 = pr1->get_vrtp_info(c1);
01406    else
01407       vp1 = NULL;
01408    if (!p0 || !p1) {
01409       /* Somebody doesn't want to play... */
01410       ast_mutex_unlock(&c0->lock);
01411       ast_mutex_unlock(&c1->lock);
01412       return -2;
01413    }
01414    if (pr0->get_codec)
01415       codec0 = pr0->get_codec(c0);
01416    else
01417       codec0 = 0;
01418    if (pr1->get_codec)
01419       codec1 = pr1->get_codec(c1);
01420    else
01421       codec1 = 0;
01422    if (pr0->get_codec && pr1->get_codec) {
01423       /* Hey, we can't do reinvite if both parties speak diffrent codecs */
01424       if (!(codec0 & codec1)) {
01425          ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, cannot native bridge.\n",codec0,codec1);
01426          ast_mutex_unlock(&c0->lock);
01427          ast_mutex_unlock(&c1->lock);
01428          return -2;
01429       }
01430    }
01431    if (pr0->set_rtp_peer(c0, p1, vp1, codec1)) 
01432       ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
01433    else {
01434       /* Store RTP peer */
01435       ast_rtp_get_peer(p1, &ac1);
01436       if (vp1)
01437          ast_rtp_get_peer(vp1, &vac1);
01438    }
01439    if (pr1->set_rtp_peer(c1, p0, vp0, codec0))
01440       ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
01441    else {
01442       /* Store RTP peer */
01443       ast_rtp_get_peer(p0, &ac0);
01444       if (vp0)
01445          ast_rtp_get_peer(vp0, &vac0);
01446    }
01447    ast_mutex_unlock(&c0->lock);
01448    ast_mutex_unlock(&c1->lock);
01449    cs[0] = c0;
01450    cs[1] = c1;
01451    cs[2] = NULL;
01452    oldcodec0 = codec0;
01453    oldcodec1 = codec1;
01454    for (;;) {
01455       if ((c0->pvt->pvt != pvt0)  ||
01456          (c1->pvt->pvt != pvt1) ||
01457          (c0->masq || c0->masqr || c1->masq || c1->masqr)) {
01458             ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
01459             if (c0->pvt->pvt == pvt0) {
01460                if (pr0->set_rtp_peer(c0, NULL, NULL, 0)) 
01461                   ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
01462             }
01463             if (c1->pvt->pvt == pvt1) {
01464                if (pr1->set_rtp_peer(c1, NULL, NULL, 0)) 
01465                   ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
01466             }
01467             /* Tell it to try again later */
01468             return -3;
01469       }
01470       to = -1;
01471       ast_rtp_get_peer(p1, &t1);
01472       ast_rtp_get_peer(p0, &t0);
01473       if (pr0->get_codec)
01474          codec0 = pr0->get_codec(c0);
01475       if (pr1->get_codec)
01476          codec1 = pr1->get_codec(c1);
01477       if (vp1)
01478          ast_rtp_get_peer(vp1, &vt1);
01479       if (vp0)
01480          ast_rtp_get_peer(vp0, &vt0);
01481       if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) {
01482          ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", 
01483             c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t1.sin_addr), ntohs(t1.sin_port), codec1);
01484          ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n", 
01485             c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vt1.sin_addr), ntohs(vt1.sin_port), codec1);
01486          ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
01487             c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
01488          ast_log(LOG_DEBUG, "Oooh, '%s' wasv %s:%d/(format %d)\n", 
01489             c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
01490          if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1)) 
01491             ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
01492          memcpy(&ac1, &t1, sizeof(ac1));
01493          memcpy(&vac1, &vt1, sizeof(vac1));
01494          oldcodec1 = codec1;
01495       }
01496       if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) {
01497          ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", 
01498             c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t0.sin_addr), ntohs(t0.sin_port), codec0);
01499          ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
01500             c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
01501          if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0))
01502             ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
01503          memcpy(&ac0, &t0, sizeof(ac0));
01504          memcpy(&vac0, &vt0, sizeof(vac0));
01505          oldcodec0 = codec0;
01506       }
01507       who = ast_waitfor_n(cs, 2, &to);
01508       if (!who) {
01509          ast_log(LOG_DEBUG, "Ooh, empty read...\n");
01510          /* check for hagnup / whentohangup */
01511          if (ast_check_hangup(c0) || ast_check_hangup(c1))
01512             break;
01513          continue;
01514       }
01515       f = ast_read(who);
01516       if (!f || ((f->frametype == AST_FRAME_DTMF) &&
01517                (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 
01518                 ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
01519          *fo = f;
01520          *rc = who;
01521          ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
01522          if ((c0->pvt->pvt == pvt0) && (!c0->_softhangup)) {
01523             if (pr0->set_rtp_peer(c0, NULL, NULL, 0)) 
01524                ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
01525          }
01526          if ((c1->pvt->pvt == pvt1) && (!c1->_softhangup)) {
01527             if (pr1->set_rtp_peer(c1, NULL, NULL, 0)) 
01528                ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
01529          }
01530          /* That's all we needed */
01531          return 0;
01532       } else {
01533          if ((f->frametype == AST_FRAME_DTMF) || 
01534             (f->frametype == AST_FRAME_VOICE) || 
01535             (f->frametype == AST_FRAME_VIDEO)) {
01536             /* Forward voice or DTMF frames if they happen upon us */
01537             if (who == c0) {
01538                ast_write(c1, f);
01539             } else if (who == c1) {
01540                ast_write(c0, f);
01541             }
01542          }
01543          ast_frfree(f);
01544       }
01545       /* Swap priority not that it's a big deal at this point */
01546       cs[2] = cs[0];
01547       cs[0] = cs[1];
01548       cs[1] = cs[2];
01549       
01550    }
01551    return -1;
01552 }

void ast_rtp_destroy (  struct ast_rtp *  rtp  ) 
 

Definition at line 958 of file rtp.c.

References ast_io_remove(), ast_smoother_free(), free, ast_rtp::io, ast_rtp::ioid, ast_rtp::rtcp, ast_rtcp::s, ast_rtp::s, and ast_rtp::smoother.

00959 {
00960    if (rtp->smoother)
00961       ast_smoother_free(rtp->smoother);
00962    if (rtp->ioid)
00963       ast_io_remove(rtp->io, rtp->ioid);
00964    if (rtp->s > -1)
00965       close(rtp->s);
00966    if (rtp->rtcp) {
00967       close(rtp->rtcp->s);
00968       free(rtp->rtcp);
00969    }
00970    free(rtp);
00971 }

int ast_rtp_fd (  struct ast_rtp *  rtp  ) 
 

Definition at line 114 of file rtp.c.

References ast_rtp::s.

00115 {
00116    return rtp->s;
00117 }

void ast_rtp_get_current_formats (  struct ast_rtp *  rtp,
int *  astFormats,
int *  nonAstFormats
) 
 

Definition at line 717 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, and rtpPayloadType::isAstFormat.

00718                                                    {
00719   int pt;
00720 
00721   *astFormats = *nonAstFormats = 0;
00722   for (pt = 0; pt < MAX_RTP_PT; ++pt) {
00723     if (rtp->current_RTP_PT[pt].isAstFormat) {
00724       *astFormats |= rtp->current_RTP_PT[pt].code;
00725     } else {
00726       *nonAstFormats |= rtp->current_RTP_PT[pt].code;
00727     }
00728   }
00729 }

void ast_rtp_get_peer (  struct ast_rtp *  rtp,
struct sockaddr_in *  them
) 
 

Definition at line 936 of file rtp.c.

References ast_rtp::them.

Referenced by ast_rtp_bridge().

00937 {
00938    them->sin_family = AF_INET;
00939    them->sin_port = rtp->them.sin_port;
00940    them->sin_addr = rtp->them.sin_addr;
00941 }

void ast_rtp_get_us (  struct ast_rtp *  rtp,
struct sockaddr_in *  us
) 
 

Definition at line 943 of file rtp.c.

References ast_rtp::us.

00944 {
00945    memcpy(us, &rtp->us, sizeof(rtp->us));
00946 }

void ast_rtp_init (  void   ) 
 

Definition at line 1598 of file rtp.c.

References ast_rtp_reload().

Referenced by main().

01599 {
01600    ast_rtp_reload();
01601 }

int ast_rtp_lookup_code (  struct ast_rtp *  rtp,
int  isAstFormat,
int  code
) 
 

Definition at line 756 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by ast_rtp_senddigit(), and ast_rtp_write().

00756                                                                         {
00757   int pt;
00758 
00759 
00760   if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
00761       code == rtp->rtp_lookup_code_cache_code) {
00762     /* Use our cached mapping, to avoid the overhead of the loop below */
00763     return rtp->rtp_lookup_code_cache_result;
00764   }
00765 
00766    /* Check the dynamic list first */
00767   for (pt = 0; pt < MAX_RTP_PT; ++pt) {
00768     if (rtp->current_RTP_PT[pt].code == code &&
00769       rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
00770       rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
00771       rtp->rtp_lookup_code_cache_code = code;
00772       rtp->rtp_lookup_code_cache_result = pt;
00773       return pt;
00774     }
00775   }
00776 
00777    /* Then the static list */
00778   for (pt = 0; pt < MAX_RTP_PT; ++pt) {
00779     if (static_RTP_PT[pt].code == code &&
00780       static_RTP_PT[pt].isAstFormat == isAstFormat) {
00781       rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
00782       rtp->rtp_lookup_code_cache_code = code;
00783       rtp->rtp_lookup_code_cache_result = pt;
00784       return pt;
00785     }
00786   }
00787   return -1;
00788 }

char* ast_rtp_lookup_mime_subtype (  int  isAstFormat,
int  code
) 
 

Definition at line 790 of file rtp.c.

References rtpPayloadType::code, and payloadType.

00790                                                              {
00791   int i;
00792 
00793   for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
00794     if (mimeTypes[i].payloadType.code == code &&
00795    mimeTypes[i].payloadType.isAstFormat == isAstFormat) {
00796       return mimeTypes[i].subtype;
00797     }
00798   }
00799   return "";
00800 }

struct rtpPayloadType ast_rtp_lookup_pt (  struct ast_rtp *  rtp,
int  pt
) 
 

Definition at line 738 of file rtp.c.

References rtpPayloadType::code, and rtpPayloadType::isAstFormat.

Referenced by ast_rtp_read().

00739 {
00740   struct rtpPayloadType result;
00741 
00742   result.isAstFormat = result.code = 0;
00743   if (pt < 0 || pt > MAX_RTP_PT) {
00744     return result; /* bogus payload type */
00745   }
00746   /* Start with the negotiated codecs */
00747   if (!rtp->rtp_offered_from_local)
00748     result = rtp->current_RTP_PT[pt];
00749   /* If it doesn't exist, check our static RTP type list, just in case */
00750   if (!result.code) 
00751     result = static_RTP_PT[pt];
00752   return result;
00753 }

struct ast_rtp* ast_rtp_new (  struct sched_context *  sched,
struct io_context *  io,
int  rtcpenable,
int  callbackmode
) 
 

Definition at line 911 of file rtp.c.

References ast_rtp_new_with_bindaddr().

00912 {
00913    struct in_addr ia;
00914    memset(&ia, 0, sizeof(ia));
00915    return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
00916 }

struct ast_rtp* ast_rtp_new_with_bindaddr (  struct sched_context *  sched,
struct io_context *  io,
int  rtcpenable,
int  callbackmode,
struct in_addr  in
) 
 

Definition at line 835 of file rtp.c.

References ast_io_add(), AST_IO_IN, ast_log(), ast_rtp_pt_default(), free, ast_rtp::io, ast_rtp::ioid, LOG_WARNING, malloc, ast_rtp::rtcp, ast_rtcp::s, ast_rtp::s, ast_rtp::sched, ast_rtp::seqno, ast_rtp::ssrc, ast_rtp::them, ast_rtcp::us, and ast_rtp::us.

Referenced by ast_rtp_new().

00836 {
00837    struct ast_rtp *rtp;
00838    int x;
00839    int first;
00840    int startplace;
00841    rtp = malloc(sizeof(struct ast_rtp));
00842    if (!rtp)
00843       return NULL;
00844    memset(rtp, 0, sizeof(struct ast_rtp));
00845    rtp->them.sin_family = AF_INET;
00846    rtp->us.sin_family = AF_INET;
00847    rtp->s = rtp_socket();
00848    rtp->ssrc = rand();
00849    rtp->seqno = rand() & 0xffff;
00850    if (rtp->s < 0) {
00851       free(rtp);
00852       ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno));
00853       return NULL;
00854    }
00855    if (sched && rtcpenable) {
00856       rtp->sched = sched;
00857       rtp->rtcp = ast_rtcp_new();
00858    }
00859    /* Find us a place */
00860    x = (rand() % (rtpend-rtpstart)) + rtpstart;
00861    x = x & ~1;
00862    startplace = x;
00863    for (;;) {
00864       /* Must be an even port number by RTP spec */
00865       rtp->us.sin_port = htons(x);
00866       rtp->us.sin_addr = addr;
00867       if (rtp->rtcp)
00868          rtp->rtcp->us.sin_port = htons(x + 1);
00869       if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
00870          (!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
00871          break;
00872       if (!first) {
00873          /* Primary bind succeeded! Gotta recreate it */
00874          close(rtp->s);
00875          rtp->s = rtp_socket();
00876       }
00877       if (errno != EADDRINUSE) {
00878          ast_log(LOG_WARNING, "Unexpected bind error: %s\n", strerror(errno));
00879          close(rtp->s);
00880          if (rtp->rtcp) {
00881             close(rtp->rtcp->s);
00882             free(rtp->rtcp);
00883          }
00884          free(rtp);
00885          return NULL;
00886       }
00887       x += 2;
00888       if (x > rtpend)
00889          x = (rtpstart + 1) & ~1;
00890       if (x == startplace) {
00891          ast_log(LOG_WARNING, "No RTP ports remaining\n");
00892          close(rtp->s);
00893          if (rtp->rtcp) {
00894             close(rtp->rtcp->s);
00895             free(rtp->rtcp);
00896          }
00897          free(rtp);
00898          return NULL;
00899       }
00900    }
00901    if (io && sched && callbackmode) {
00902       /* Operate this one in a callback mode */
00903       rtp->sched = sched;
00904       rtp->io = io;
00905       rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
00906    }
00907    ast_rtp_pt_default(rtp);
00908    return rtp;
00909 }

void ast_rtp_offered_from_local (  struct ast_rtp *  rtp,
int  local
) 
 

Definition at line 731 of file rtp.c.

References ast_log(), LOG_WARNING, and ast_rtp::rtp_offered_from_local.

00731                                                                 {
00732   if (rtp)
00733     rtp->rtp_offered_from_local = local;
00734   else
00735     ast_log(LOG_WARNING, "rtp structure is null\n");
00736 }

int ast_rtp_proto_register (  struct ast_rtp_protocol *  proto  ) 
 

Definition at line 1322 of file rtp.c.

References ast_log(), LOG_WARNING, ast_rtp_protocol::next, and ast_rtp_protocol::type.

01323 {
01324    struct ast_rtp_protocol *cur;
01325    cur = protos;
01326    while(cur) {
01327       if (cur->type == proto->type) {
01328          ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
01329          return -1;
01330       }
01331       cur = cur->next;
01332    }
01333    proto->next = protos;
01334    protos = proto;
01335    return 0;
01336 }

void ast_rtp_proto_unregister (  struct ast_rtp_protocol *  proto  ) 
 

Definition at line 1304 of file rtp.c.

References ast_rtp_protocol::next.

01305 {
01306    struct ast_rtp_protocol *cur, *prev;
01307    cur = protos;
01308    prev = NULL;
01309    while(cur) {
01310       if (cur == proto) {
01311          if (prev)
01312             prev->next = proto->next;
01313          else
01314             protos = proto->next;
01315          return;
01316       }
01317       prev = cur;
01318       cur = cur->next;
01319    }
01320 }

void ast_rtp_pt_clear (  struct ast_rtp *  rtp  ) 
 

Definition at line 659 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

00660 {
00661   int i;
00662 
00663   for (i = 0; i < MAX_RTP_PT; ++i) {
00664     rtp->current_RTP_PT[i].isAstFormat = 0;
00665     rtp->current_RTP_PT[i].code = 0;
00666   }
00667 
00668   rtp->rtp_lookup_code_cache_isAstFormat = 0;
00669   rtp->rtp_lookup_code_cache_code = 0;
00670   rtp->rtp_lookup_code_cache_result = 0;
00671 }

void ast_rtp_pt_default (  struct ast_rtp *  rtp  ) 
 

Definition at line 673 of file rtp.c.

References rtpPayloadType::code, ast_rtp::current_RTP_PT, rtpPayloadType::isAstFormat, ast_rtp::rtp_lookup_code_cache_code, ast_rtp::rtp_lookup_code_cache_isAstFormat, and ast_rtp::rtp_lookup_code_cache_result.

Referenced by ast_rtp_new_with_bindaddr().

00674 {
00675   int i;
00676   /* Initialize to default payload types */
00677   for (i = 0; i < MAX_RTP_PT; ++i) {
00678     rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
00679     rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
00680   }
00681 
00682   rtp->rtp_lookup_code_cache_isAstFormat = 0;
00683   rtp->rtp_lookup_code_cache_code = 0;
00684   rtp->rtp_lookup_code_cache_result = 0;
00685 }

struct ast_frame* ast_rtp_read (  struct ast_rtp *  rtp  ) 
 

Definition at line 401 of file rtp.c.

References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_MAX_AUDIO, AST_FORMAT_SLINEAR, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, ast_frame_byteswap_be, AST_FRAME_NULL, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRIENDLY_OFFSET, ast_getformatname(), ast_inet_ntoa(), ast_log(), AST_RTP_CISCO_DTMF, AST_RTP_CN, AST_RTP_DTMF, ast_rtp_lookup_pt(), rtpPayloadType::code, CRASH, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp::dtmfcount, ast_rtp::f, ast_frame::frametype, rtpPayloadType::isAstFormat, ast_rtp::lasteventseqn, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, ast_rtp::lastrxts, LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, ast_rtp::nat, ast_frame::offset, ast_rtp::rawdata, ast_rtp::resp, ast_rtp::s, ast_frame::samples, ast_frame::src, ast_frame::subclass, and ast_rtp::them.

00402 {
00403    int res;
00404    struct sockaddr_in sin;
00405    int len;
00406    unsigned int seqno;
00407    int version;
00408    int payloadtype;
00409    int hdrlen = 12;
00410    int padding;
00411    int mark;
00412    int ext;
00413    char iabuf[INET_ADDRSTRLEN];
00414    unsigned int timestamp;
00415    unsigned int *rtpheader;
00416    static struct ast_frame *f, null_frame = { AST_FRAME_NULL, };
00417    struct rtpPayloadType rtpPT;
00418    
00419    len = sizeof(sin);
00420    
00421    /* Cache where the header will go */
00422    res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
00423                0, (struct sockaddr *)&sin, &len);
00424 
00425 
00426    rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
00427    if (res < 0) {
00428       if (errno != EAGAIN)
00429          ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
00430       if (errno == EBADF)
00431          CRASH;
00432       return &null_frame;
00433    }
00434    if (res < hdrlen) {
00435       ast_log(LOG_WARNING, "RTP Read too short\n");
00436       return &null_frame;
00437    }
00438 
00439    /* Ignore if the other side hasn't been given an address
00440       yet.  */
00441    if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
00442       return &null_frame;
00443 
00444    if (rtp->nat) {
00445       /* Send to whoever sent to us */
00446       if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
00447           (rtp->them.sin_port != sin.sin_port)) {
00448          memcpy(&rtp->them, &sin, sizeof(rtp->them));
00449          ast_log(LOG_DEBUG, "RTP NAT: Using address %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
00450       }
00451    }
00452 
00453    /* Get fields */
00454    seqno = ntohl(rtpheader[0]);
00455 
00456    /* Check RTP version */
00457    version = (seqno & 0xC0000000) >> 30;
00458    if (version != 2)
00459       return &null_frame;
00460    
00461    payloadtype = (seqno & 0x7f0000) >> 16;
00462    padding = seqno & (1 << 29);
00463    mark = seqno & (1 << 23);
00464    ext = seqno & (1 << 28);
00465    seqno &= 0xffff;
00466    timestamp = ntohl(rtpheader[1]);
00467    
00468    if (padding) {
00469       /* Remove padding bytes */
00470       res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
00471    }
00472    
00473    if (ext) {
00474       /* RTP Extension present */
00475       hdrlen += 4;
00476       hdrlen += (ntohl(rtpheader[3]) & 0xffff) << 2;
00477    }
00478 
00479    if (res < hdrlen) {
00480       ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
00481       return &null_frame;
00482    }
00483 
00484 #if 0
00485    printf("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len = %d)\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
00486 #endif   
00487    rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
00488    if (!rtpPT.isAstFormat) {
00489      /* This is special in-band data that's not one of our codecs */
00490      if (rtpPT.code == AST_RTP_DTMF) {
00491        /* It's special -- rfc2833 process it */
00492        if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) {
00493          f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
00494          rtp->lasteventseqn = seqno;
00495        } else f = NULL;
00496        if (f) return f; else return &null_frame;
00497      } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
00498        /* It's really special -- process it the Cisco way */
00499        if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) {
00500          f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
00501          rtp->lasteventseqn = seqno;
00502        } else f = NULL;
00503        if (f) return f; else return &null_frame;
00504      } else if (rtpPT.code == AST_RTP_CN) {
00505        /* Comfort Noise */
00506        f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
00507        if (f) return f; else return &null_frame;
00508      } else {
00509        ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
00510        return &null_frame;
00511      }
00512    }
00513    rtp->f.subclass = rtpPT.code;
00514    if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO)
00515       rtp->f.frametype = AST_FRAME_VOICE;
00516    else
00517       rtp->f.frametype = AST_FRAME_VIDEO;
00518    rtp->lastrxformat = rtp->f.subclass;
00519 
00520    if (!rtp->lastrxts)
00521       rtp->lastrxts = timestamp;
00522 
00523    if (rtp->dtmfcount) {
00524 #if 0
00525       printf("dtmfcount was %d\n", rtp->dtmfcount);
00526 #endif      
00527       rtp->dtmfcount -= (timestamp - rtp->lastrxts);
00528       if (rtp->dtmfcount < 0)
00529          rtp->dtmfcount = 0;
00530 #if 0
00531       if (dtmftimeout != rtp->dtmfcount)
00532          printf("dtmfcount is %d\n", rtp->dtmfcount);
00533 #endif
00534    }
00535    rtp->lastrxts = timestamp;
00536 
00537    /* Send any pending DTMF */
00538    if (rtp->resp && !rtp->dtmfcount) {
00539       ast_log(LOG_DEBUG, "Sending pending DTMF\n");
00540       return send_dtmf(rtp);
00541    }
00542    rtp->f.mallocd = 0;
00543    rtp->f.datalen = res - hdrlen;
00544    rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
00545    rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
00546    if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
00547       switch(rtp->f.subclass) {
00548       case AST_FORMAT_ULAW:
00549       case AST_FORMAT_ALAW:
00550          rtp->f.samples = rtp->f.datalen;
00551          break;
00552       case AST_FORMAT_SLINEAR:
00553          rtp->f.samples = rtp->f.datalen / 2;
00554          ast_frame_byteswap_be(&rtp->f);
00555          break;
00556       case AST_FORMAT_GSM:
00557          rtp->f.samples = 160 * (rtp->f.datalen / 33);
00558          break;
00559       case AST_FORMAT_ILBC:
00560          rtp->f.samples = 240 * (rtp->f.datalen / 50);
00561          break;
00562       case AST_FORMAT_ADPCM:
00563       case AST_FORMAT_G726:
00564          rtp->f.samples = rtp->f.datalen * 2;
00565          break;
00566       case AST_FORMAT_G729A:
00567          rtp->f.samples = rtp->f.datalen * 8;
00568          break;
00569       case AST_FORMAT_G723_1:
00570          rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen);
00571          break;
00572       case AST_FORMAT_SPEEX:
00573          /* assumes that the RTP packet contained one Speex frame */
00574            rtp->f.samples = 160;
00575          break;
00576       case AST_FORMAT_LPC10:
00577           rtp->f.samples = 22 * 8;
00578          rtp->f.samples += (((char *)(rtp->f.data))[7] & 0x1) * 8;
00579          break;
00580       default:
00581          ast_log(LOG_NOTICE, "Unable to calculate samples for format %s\n", ast_getformatname(rtp->f.subclass));
00582          break;
00583       }
00584       calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
00585    } else {
00586       /* Video -- samples is # of samples vs. 90000 */
00587       if (!rtp->lastividtimestamp)
00588          rtp->lastividtimestamp = timestamp;
00589       rtp->f.samples = timestamp - rtp->lastividtimestamp;
00590       rtp->lastividtimestamp = timestamp;
00591       rtp->f.delivery.tv_sec = 0;
00592       rtp->f.delivery.tv_usec = 0;
00593       if (mark)
00594          rtp->f.subclass |= 0x1;
00595       
00596    }
00597    rtp->f.src = "RTP";
00598    return &rtp->f;
00599 }

void ast_rtp_reload (  void   ) 
 

Definition at line 1554 of file rtp.c.

References ast_destroy(), ast_false(), ast_load(), ast_log(), ast_variable_retrieve(), ast_verbose(), LOG_WARNING, option_verbose, and VERBOSE_PREFIX_2.

Referenced by ast_module_reload(), ast_rtp_init(), and main().

01555 {
01556    struct ast_config *cfg;
01557    char *s;
01558    rtpstart = 5000;
01559    rtpend = 31000;
01560    cfg = ast_load("rtp.conf");
01561    if (cfg) {
01562       if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
01563          rtpstart = atoi(s);
01564          if (rtpstart < 1024)
01565             rtpstart = 1024;
01566          if (rtpstart > 65535)
01567             rtpstart = 65535;
01568       }
01569       if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
01570          rtpend = atoi(s);
01571          if (rtpend < 1024)
01572             rtpend = 1024;
01573          if (rtpend > 65535)
01574             rtpend = 65535;
01575       }
01576       if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
01577 #ifdef SO_NO_CHECK
01578          if (ast_false(s))
01579             nochecksums = 1;
01580          else
01581             nochecksums = 0;
01582 #else
01583          if (ast_false(s))
01584             ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
01585 #endif
01586       }
01587       ast_destroy(cfg);
01588    }
01589    if (rtpstart >= rtpend) {
01590       ast_log(LOG_WARNING, "Unreasonable values for RTP start/end\n");
01591       rtpstart = 5000;
01592       rtpend = 31000;
01593    }
01594    if (option_verbose > 1)
01595       ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
01596 }

int ast_rtp_senddigit (  struct ast_rtp *  rtp,
char  digit
) 
 

Definition at line 999 of file rtp.c.

References ast_inet_ntoa(), ast_log(), AST_RTP_DTMF, ast_rtp_lookup_code(), ast_rtp::dtmfmute, ast_rtp::lastts, LOG_NOTICE, LOG_WARNING, ast_rtp::s, ast_rtp::seqno, ast_rtp::ssrc, and ast_rtp::them.

01000 {
01001    unsigned int *rtpheader;
01002    int hdrlen = 12;
01003    int res;
01004    int x;
01005    int payload;
01006    char data[256];
01007    char iabuf[INET_ADDRSTRLEN];
01008 
01009    if ((digit <= '9') && (digit >= '0'))
01010       digit -= '0';
01011    else if (digit == '*')
01012       digit = 10;
01013    else if (digit == '#')
01014       digit = 11;
01015    else if ((digit >= 'A') && (digit <= 'D')) 
01016       digit = digit - 'A' + 12;
01017    else if ((digit >= 'a') && (digit <= 'd')) 
01018       digit = digit - 'a' + 12;
01019    else {
01020       ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
01021       return -1;
01022    }
01023    payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
01024 
01025    /* If we have no peer, return immediately */ 
01026    if (!rtp->them.sin_addr.s_addr)
01027       return 0;
01028 
01029    gettimeofday(&rtp->dtmfmute, NULL);
01030    rtp->dtmfmute.tv_usec += (500 * 1000);
01031    if (rtp->dtmfmute.tv_usec > 1000000) {
01032       rtp->dtmfmute.tv_usec -= 1000000;
01033       rtp->dtmfmute.tv_sec += 1;
01034    }
01035    
01036    /* Get a pointer to the header */
01037    rtpheader = (unsigned int *)data;
01038    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
01039    rtpheader[1] = htonl(rtp->lastts);
01040    rtpheader[2] = htonl(rtp->ssrc); 
01041    rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
01042    for (x=0;x<6;x++) {
01043       if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
01044          res = sendto(rtp->s, (void *)rtpheader, hdrlen + 4, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
01045          if (res <0) 
01046             ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
01047    #if 0
01048       printf("Sent %d bytes of RTP data to %s:%d\n", res, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
01049    #endif      
01050       }
01051       if (x == 2) {
01052          /* Clear marker bit and increment seqno */
01053          rtpheader[0] = htonl((2 << 30)  | (payload << 16) | (rtp->seqno++));
01054          /* Make duration 800 (100ms) */
01055          rtpheader[3] |= htonl((800));
01056          /* Set the End bit for the last 3 */
01057          rtpheader[3] |= htonl((1 << 23));
01058       } else if ( x < 5) {
01059          rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno++));
01060       }
01061    }
01062    return 0;
01063 }

void ast_rtp_set_callback (  struct ast_rtp *  rtp,
ast_rtp_callback  callback
) 
 

Definition at line 167 of file rtp.c.

References ast_rtp::callback.

00168 {
00169    rtp->callback = callback;
00170 }

void ast_rtp_set_data (  struct ast_rtp *  rtp,
void *  data
) 
 

Definition at line 162 of file rtp.c.

References ast_rtp::data.

00163 {
00164    rtp->data = data;
00165 }

void ast_rtp_set_m_type (  struct ast_rtp *  rtp,
int  pt
) 
 

Definition at line 690 of file rtp.c.

References rtpPayloadType::code, and ast_rtp::current_RTP_PT.

00690                                                      {
00691   if (pt < 0 || pt > MAX_RTP_PT) return; /* bogus payload type */
00692 
00693   if (static_RTP_PT[pt].code != 0) {
00694     rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
00695   }
00696 }

void ast_rtp_set_peer (  struct ast_rtp *  rtp,
struct sockaddr_in *  them
) 
 

Definition at line 926 of file rtp.c.

References ast_rtp::rtcp, ast_rtcp::them, and ast_rtp::them.

00927 {
00928    rtp->them.sin_port = them->sin_port;
00929    rtp->them.sin_addr = them->sin_addr;
00930    if (rtp->rtcp) {
00931       rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
00932       rtp->rtcp->them.sin_addr = them->sin_addr;
00933    }
00934 }

void ast_rtp_set_rtpmap_type (  struct ast_rtp *  rtp,
int  pt,
char *  mimeType,
char *  mimeSubtype
) 
 

Definition at line 700 of file rtp.c.

References ast_rtp::current_RTP_PT, subtype, and type.

00701                                              {
00702   int i;
00703 
00704   if (pt < 0 || pt > MAX_RTP_PT) return; /* bogus payload type */
00705 
00706   for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
00707     if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
00708    strcasecmp(mimeType, mimeTypes[i].type) == 0) {
00709       rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
00710       return;
00711     }
00712   }
00713 }

void ast_rtp_setnat (  struct ast_rtp *  rtp,
int  nat
) 
 

Definition at line 172 of file rtp.c.

References ast_rtp::nat.

00173 {
00174    rtp->nat = nat;
00175 }

int ast_rtp_settos (  struct ast_rtp *  rtp,
int  tos
) 
 

Definition at line 918 of file rtp.c.

References ast_log(), LOG_WARNING, and ast_rtp::s.

00919 {
00920    int res;
00921    if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
00922       ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
00923    return res;
00924 }

void ast_rtp_stop (  struct ast_rtp *  rtp  ) 
 

Definition at line 948 of file rtp.c.

References ast_rtp::rtcp, ast_rtcp::them, and ast_rtp::them.

00949 {
00950    memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
00951    memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
00952    if (rtp->rtcp) {
00953       memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
00954       memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->them.sin_port));
00955    }
00956 }

int ast_rtp_write (  struct ast_rtp *  rtp,
struct ast_frame *  f
) 
 

Definition at line 1161 of file rtp.c.

References AST_FORMAT_ADPCM, AST_FORMAT_ALAW, AST_FORMAT_G723_1, AST_FORMAT_G726, AST_FORMAT_G729A, AST_FORMAT_GSM, AST_FORMAT_H261, AST_FORMAT_H263, AST_FORMAT_ILBC, AST_FORMAT_LPC10, AST_FORMAT_SLINEAR, AST_FORMAT_SPEEX, AST_FORMAT_ULAW, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frdup(), ast_getformatname(), ast_log(), ast_rtp_lookup_code(), ast_smoother_feed, ast_smoother_feed_be, AST_SMOOTHER_FLAG_G729, ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_smoother_set_flags(), ast_frame::datalen, ast_frame::frametype, ast_rtp::lasttxformat, LOG_DEBUG, LOG_WARNING, ast_frame::offset, ast_rtp::smoother, ast_frame::subclass, and ast_rtp::them.

01162 {
01163    struct ast_frame *f;
01164    int codec;
01165    int hdrlen = 12;
01166    int subclass;
01167    
01168 
01169    /* If we have no peer, return immediately */ 
01170    if (!rtp->them.sin_addr.s_addr)
01171       return 0;
01172 
01173    /* If there is no data length, return immediately */
01174    if (!_f->datalen) 
01175       return 0;
01176    
01177    /* Make sure we have enough space for RTP header */
01178    if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
01179       ast_log(LOG_WARNING, "RTP can only send voice\n");
01180       return -1;
01181    }
01182 
01183    subclass = _f->subclass;
01184    if (_f->frametype == AST_FRAME_VIDEO)
01185       subclass &= ~0x1;
01186 
01187    codec = ast_rtp_lookup_code(rtp, 1, subclass);
01188    if (codec < 0) {
01189       ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
01190       return -1;
01191    }
01192 
01193    if (rtp->lasttxformat != subclass) {
01194       /* New format, reset the smoother */
01195       ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
01196       rtp->lasttxformat = subclass;
01197       if (rtp->smoother)
01198          ast_smoother_free(rtp->smoother);
01199       rtp->smoother = NULL;
01200    }
01201 
01202 
01203    switch(subclass) {
01204    case AST_FORMAT_SLINEAR:
01205       if (!rtp->smoother) {
01206          rtp->smoother = ast_smoother_new(320);
01207       }
01208       if (!rtp->smoother) {
01209          ast_log(LOG_WARNING, "Unable to create smoother :(\n");
01210          return -1;
01211       }
01212       ast_smoother_feed_be(rtp->smoother, _f);
01213       
01214       while((f = ast_smoother_read(rtp->smoother)))
01215          ast_rtp_raw_write(rtp, f, codec);
01216       break;
01217    case AST_FORMAT_ULAW:
01218    case AST_FORMAT_ALAW:
01219       if (!rtp->smoother) {
01220          rtp->smoother = ast_smoother_new(160);
01221       }
01222       if (!rtp->smoother) {
01223          ast_log(LOG_WARNING, "Unable to create smoother :(\n");
01224          return -1;
01225       }
01226       ast_smoother_feed(rtp->smoother, _f);
01227       
01228       while((f = ast_smoother_read(rtp->smoother)))
01229          ast_rtp_raw_write(rtp, f, codec);
01230       break;
01231    case AST_FORMAT_ADPCM:
01232    case AST_FORMAT_G726:
01233       if (!rtp->smoother) {
01234          rtp->smoother = ast_smoother_new(80);
01235       }
01236       if (!rtp->smoother) {
01237          ast_log(LOG_WARNING, "Unable to create smoother :(\n");
01238          return -1;
01239       }
01240       ast_smoother_feed(rtp->smoother, _f);
01241       
01242       while((f = ast_smoother_read(rtp->smoother)))
01243          ast_rtp_raw_write(rtp, f, codec);
01244       break;
01245    case AST_FORMAT_G729A:
01246       if (!rtp->smoother) {
01247          rtp->smoother = ast_smoother_new(20);
01248          if (rtp->smoother)
01249             ast_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_G729);
01250       }
01251       if (!rtp->smoother) {
01252          ast_log(LOG_WARNING, "Unable to create g729 smoother :(\n");
01253          return -1;
01254       }
01255       ast_smoother_feed(rtp->smoother, _f);
01256       
01257       while((f = ast_smoother_read(rtp->smoother)))
01258          ast_rtp_raw_write(rtp, f, codec);
01259       break;
01260    case AST_FORMAT_GSM:
01261       if (!rtp->smoother) {
01262          rtp->smoother = ast_smoother_new(33);
01263       }
01264       if (!rtp->smoother) {
01265          ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n");
01266          return -1;
01267       }
01268       ast_smoother_feed(rtp->smoother, _f);
01269       while((f = ast_smoother_read(rtp->smoother)))
01270          ast_rtp_raw_write(rtp, f, codec);
01271       break;
01272    case AST_FORMAT_ILBC:
01273       if (!rtp->smoother) {
01274          rtp->smoother = ast_smoother_new(50);
01275       }
01276       if (!rtp->smoother) {
01277          ast_log(LOG_WARNING, "Unable to create ILBC smoother :(\n");
01278          return -1;
01279       }
01280       ast_smoother_feed(rtp->smoother, _f);
01281       while((f = ast_smoother_read(rtp->smoother)))
01282          ast_rtp_raw_write(rtp, f, codec);
01283       break;
01284    default: 
01285       ast_log(LOG_WARNING, "Not sure about sending format %s packets\n", ast_getformatname(subclass));
01286       /* fall through to... */
01287    case AST_FORMAT_H261:
01288    case AST_FORMAT_H263:
01289    case AST_FORMAT_G723_1:
01290    case AST_FORMAT_LPC10:
01291    case AST_FORMAT_SPEEX:
01292            /* Don't buffer outgoing frames; send them one-per-packet: */
01293       if (_f->offset < hdrlen) {
01294          f = ast_frdup(_f);
01295       } else {
01296          f = _f;
01297       }
01298       ast_rtp_raw_write(rtp, f, codec);
01299    }
01300       
01301    return 0;
01302 }

Generated on Wed Aug 10 11:36:34 2005 for Asterisk by  doxygen 1.4.4