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app_rtsp_sip.c
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app_rtsp_sip.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Sergio Garcia Murillo <sergio.garcia@fontventa.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief MP4 application -- save and play mp4 files
*
* \ingroup applications
*/
/*
* Tommy Long
* Port to Asterisk version 17.3 and 17.5.
*
* Port 17.3 upgrades this code to run on Asterisk version 17.3, and adds
* a thin SIP protocol in order to send audio to the same target device.
* The development environment is Ubuntu 16.04 Desktop on 32bit machine.
* Porting Generals:
* malloc() is now ast_malloc()
* free() is now ast_free()
* strdup() is now ast_strdup
* strndup() is now ast_strndup()
* memcpy() of overlapping memory is being changed to memmove().
* This code was using memcpy to copy overlapping memory which is prone to errors (as I found out).
* memmove() is recommended instead.
* ast_log(LOG_DEBUG,....) change to ast_debug(<level>,...)
*
* SIP is added for sending a one-way call to the same end device.
* The SIP protocol used is very simple and lite and completely self-contained within this module.
* (The exception is that it uses PJSIP's digest authentication).
* There will be for sure various aspects of SIP not supported.
*
* Documentation: in-line documentation is added.
*
* Port 17.5
* Development environment is Ubuntu 20.04 server cloud image running on KVM on 64 bit machine.
* Fixes a few minor compiler warnings that show up in newer Ubuntu.
* Primariy fixes a problem with the ast_frame and buffer structure
* used for receiving RTP from end device (using RTSP).
*
* Has not been tested for:
* 1) IPv6
* 2) RTSP Digest Authentication (newly added).
* 3) RTSP Tunnel
* 4) Use DTMF to stop RTSP.
*/
/* Use the following to test for Buffer length issues */
/* #define TEST_BUFFER */
#include <asterisk.h>
#include "asterisk/app.h" /* PORT 17.3 ADDed for parsing args */
#include <stdlib.h>
#include <stdio.h>
#include <unistd.h>
#include <string.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <errno.h>
#include <arpa/inet.h> /* NEW. needed for getsockname() */
#include <asterisk/lock.h>
#include <asterisk/file.h>
#include <asterisk/logger.h>
#include <asterisk/channel.h>
#include <asterisk/pbx.h>
#include <asterisk/module.h>
#include <asterisk/utils.h>
#include <asterisk/translate.h>
#include <asterisk/format_compatibility.h>
/* For use with pjsip digest authentication */
#include <pjsua-lib/pjsua.h>
#include <pj/string.h>
/*** DOCUMENTATION
<application name="RTSP-SIP" language="en_US">
<synopsis>
<para>Attempt to connect to another device/endpoint using RTSP
and play streaming audio (video has not been tested). </para>
<para>If commanded, will also attempt to connect to the same device/endpoint
using SIP and send an audio stream to it. </para>
</synopsis>
<description>
<para>This 'endpoint' application is intended to be used as an execution
step of an extension in a Dialplan. When executed, the application will
first attempt to contact and authenticate with the specified target device using RTSP,
gather the video and audio media types supported by the target device and match them
with the media types supported by the Asterisk channel connecting to this application.
It will choose the "best" media types for audio as well as video (if any) and play them
using RTP into the Asterisk channel and consequently to the calling party.</para>
<para>Once an audio stream is identifed and played using RTSP,
if sip has been specifically enabled, it will next attempt to contact the device using SIP.
Once contacted, it will setup an audio stream from the Asterisk channel (that is connected to
this application) to the target device using the same audio media chosen by RTSP.
Consequently it will play audio from the calling party to the target device using RTP.
This application uses its own 'lite' version of SIP and is intended
for simple uses cases, namely connecting this application as a sip UA
directly to the target device over a local area network.</para>
<example title="Dial extension 103 to talk two-way with a surveillance camera">
exten = 103,1,Answer()
same = n,Wait(1)
same = n,RTSP-SIP(rtsp://2001:mypasswd@192.168.0.34:554/live.sdp,1,streaming_server,5060)
same = n,Wait(5)
same = n,Hangup()
</example>
<example title="Dial extension 101 to receive audio only from a surveillance camera">
exten = 101,1,Answer()
same = n,Wait(1)
same = n,RTSP-SIP(rtsp://user:mypasswd@192.168.0.34:554/live.sdp,0)
same = n,Wait(5)
same = n,Hangup()
</example>
</description>
<syntax>
<parameter name="RTSP-URL" required="true">
<para>RTSP url of the target device to be played.
Format: rtsp://username:password@address/stream-id.
'username' and 'password' are optional for login to the device using Basic Authentication.
'address' is domain name or ip address of target device.
'stream-id' optionally identifies the stream to be played.</para>
</parameter>
<parameter name="enable-sip" required="true">
<para>0 if using this only to play RTSP streams.
1 if an audio stream is also to be setup to the target device.</para>
</parameter>
<parameter name="realm" required="false">
<para>If enable-sip = 1, this parameter is required.
SIP uses Digest Authentication, and uses the supplied
realm string as part of the authentication.</para>
</parameter>
<parameter name="port" required="false">
<para>If enable-sip = 1, this optional parameter can be used
to specify a different SIP port for the target device to listen on. Default is 5060. </para>
</parameter>
</syntax>
<see-also>
<ref type="application">Dial</ref>
</see-also>
</application>
***/
/* NEW SIP: messaging sip:MY_NAME@blah_blah */
#define MY_NAME "agbell"
/* PORT 17.3
* Adaptive MultiRate (AMR) Narrow Band (IETF RFC4867) is no longer supported in Asterisk.
* This file is using it to determine MIME type and
* for some unknown reason is ORing this to determine best audio formats.
* For now just zero it out .
*/
#ifndef AST_FORMAT_AMRNB
/* #define AST_FORMAT_AMRNB (1 << 13) OLD */
#define AST_FORMAT_AMRNB 0
#endif
/* PORT 17.3 AST_FORMAT_MPEG4 is now AST_FORMAT_MP4 (1LL << 22). The following longer needed. */
/* #ifndef AST_FORMAT_MPEG4 OLD
* #define AST_FORMAT_MPEG4 (1 << 22) OLD
* #endif OLD
*/
/* PORT17.3. Update to use xml based loading and documentation */
/* static char *name_rtsp_sip = "rtsp_sip"; OLD */
/* static char *syn_rtsp_sip = "sip caller with rtsp player"; */
/* static char *des_rtsp_sip = " rtsp(url): Play url. \n"; */
static const char app[] = "RTSP-SIP";
/* RTSP states */
#define RTSP_NONE 0
#define RTSP_DESCRIBE 1
#define RTSP_SETUP_AUDIO 2
#define RTSP_SETUP_VIDEO 3
#define RTSP_PLAY 4
#define RTSP_PLAYING 5
#define RTSP_RELEASED 6
/* NEW SIP states */
#define SIP_STATE_NONE 0
#define SIP_STATE_OPTIONS 1
#define SIP_STATE_INVITE 2
#define SIP_STATE_ACK 3
#define SIP_STATE_CANCEL 4
#define SIP_STATE_BYE 5
#define SIP_STATE_REFER 6
#define SIP_STATE_NOTIFY 7
#define SIP_STATE_MESSAGE 8
#define SIP_STATE_SUBSCRIBE 9
#define SIP_STATE_INFO 10
#define PKT_PAYLOAD 1450
#define PKT_SIZE (sizeof(struct ast_frame) + AST_FRIENDLY_OFFSET + PKT_PAYLOAD)
#define PKT_OFFSET (sizeof(struct ast_frame) + AST_FRIENDLY_OFFSET)
/* PORT 17.3
* Some of the following AST_FORMAT_xx were tweaked in format_compatibility.h;
* Plus the bit list is now 64bits
*/
static struct
{
/* int format; OLD */
uint64_t format;
char* name;
} mimeTypes[] = {
{ AST_FORMAT_G723, "G723"}, /* OLD { AST_FORMAT_G723_1, "G723"}, */
{ AST_FORMAT_GSM, "GSM"},
{ AST_FORMAT_ULAW, "PCMU"},
{ AST_FORMAT_ALAW, "PCMA"},
{ AST_FORMAT_G726, "G726-32"},
{ AST_FORMAT_ADPCM, "DVI4"},
{ AST_FORMAT_SLIN, "L16"}, /* OLD { AST_FORMAT_SLINEAR, "L16"}, */
{ AST_FORMAT_LPC10, "LPC"},
{ AST_FORMAT_G729, "G729"}, /* { AST_FORMAT_G729A, "G729"}, */
{ AST_FORMAT_SPEEX, "speex"},
{ AST_FORMAT_ILBC, "iLBC"},
{ AST_FORMAT_G722, "G722"},
{ AST_FORMAT_G726_AAL2, "AAL2-G726-32"},
{ AST_FORMAT_AMRNB, "AMR"},
{ AST_FORMAT_JPEG, "JPEG"},
{ AST_FORMAT_PNG, "PNG"},
{ AST_FORMAT_H261, "H261"},
{ AST_FORMAT_H263, "H263"},
/*{ AST_FORMAT_H263_PLUS, "H263-1998"}, OLD removed*/
{ AST_FORMAT_H263P, "H263-2000"},/* OLD { AST_FORMAT_H263_PLUS, "H263-2000"}, */
{ AST_FORMAT_H264, "H264"},
/* { AST_FORMAT_MPEG4, "MP4V-ES"}, OLD */
{ AST_FORMAT_MP4, "MP4V-ES"},
};
typedef enum
{
RTCP_SR = 200,
RTCP_RR = 201,
RTCP_SDES = 202,
RTCP_BYE = 203,
RTCP_APP = 204
} RtcpType;
typedef enum
{
RTCP_SDES_END = 0,
RTCP_SDES_CNAME = 1,
RTCP_SDES_NAME = 2,
RTCP_SDES_EMAIL = 3,
RTCP_SDES_PHONE = 4,
RTCP_SDES_LOC = 5,
RTCP_SDES_TOOL = 6,
RTCP_SDES_NOTE = 7,
RTCP_SDES_PRIV = 8,
RTCP_SDES_IMG = 9,
RTCP_SDES_DOOR = 10,
RTCP_SDES_SOURCE = 11
} RtcpSdesType;
struct RtcpCommonHeader
{
unsigned short count:5; /* varies by payload type */
unsigned short p:1; /* padding flag */
unsigned short version:2; /* protocol version */
unsigned short pt:8; /* payload type */
unsigned short length; /* packet length in words, without this word */
};
struct RtcpReceptionReport
{
unsigned int ssrc; /* data source being reported */
unsigned int fraction:8; /* fraction lost since last SR/RR */
int lost:24; /* cumulative number of packets lost (signed!) */
unsigned int last_seq; /* extended last sequence number received */
unsigned int jitter; /* interarrival jitter */
unsigned int lsr; /* last SR packet from this source */
unsigned int dlsr; /* delay since last SR packet */
};
struct RtcpSdesItem
{
unsigned char type; /* type of SDES item (rtcp_sdes_type_t) */
unsigned char length; /* length of SDES item (in octets) */
char data[1]; /* text, not zero-terminated */
};
struct Rtcp
{
struct RtcpCommonHeader common; /* common header */
union
{
/* sender report (SR) */
struct
{
unsigned int ssrc; /* source this RTCP packet refers to */
unsigned int ntp_sec; /* NTP timestamp */
unsigned int ntp_frac;
unsigned int rtp_ts; /* RTP timestamp */
unsigned int psent; /* packets sent */
unsigned int osent; /* octets sent */
/* variable-length list */
struct RtcpReceptionReport rr[1];
} sr;
/* reception report (RR) */
struct
{
unsigned int ssrc; /* source this generating this report */
/* variable-length list */
struct RtcpReceptionReport rr[1];
} rr;
/* BYE */
struct
{
unsigned int src[1]; /* list of sources */
/* can't express trailing text */
} bye;
/* source description (SDES) */
struct rtcp_sdes_t
{
unsigned int src; /* first SSRC/CSRC */
struct RtcpSdesItem item[1]; /* list of SDES items */
} sdes;
} r;
};
struct RtpHeader
{
unsigned int cc:4; /* CSRC count */
unsigned int x:1; /* header extension flag */
unsigned int p:1; /* padding flag */
unsigned int version:2; /* protocol version */
unsigned int pt:7; /* payload type */
unsigned int m:1; /* marker bit */
unsigned int seq:16; /* sequence number */
unsigned int ts; /* timestamp */
unsigned int ssrc; /* synchronization source */
/* unsigned int csrc[1]; * optional CSRC list. REMOVE. Not supported BY SIP. */
};
struct MediaStats
{
unsigned int count;
unsigned int minSN;
unsigned int maxSN;
unsigned int lastTS;
unsigned int ssrc;
struct timeval time;
};
static void MediaStatsReset(struct MediaStats *stats)
{
stats->count = 0;
stats->minSN = 0;
stats->maxSN = 0;
stats->lastTS = 0;
stats->time = ast_tvnow();
}
static void MediaStatsUpdate(struct MediaStats *stats,unsigned int ts,unsigned int sn,unsigned int ssrc)
{
stats->ssrc = ssrc;
stats->count++;
if (!stats->minSN)
stats->minSN = sn;
if (stats->maxSN<sn)
stats->maxSN = sn;
stats->lastTS = ts;
}
static void MediaStatsRR(struct MediaStats *stats, struct Rtcp *rtcp)
{
/* Set pointer as ssrc */
/* COMMENT
* Build a Receiver Report packet.
* An RR RTCP packet starts with the common header followed
* by the SSRC assigned to this receiver followed by report blocks
* of sender1 and sender2.
*/
/*
* COMMENT
* The next line originally set the SSRC to a pointer's value
* because the pointer value is fairly random as the SSRC value.
* However it doesn't always port/compile very well. Let's use random() instead.
*/
/* rtcp->r.rr.ssrc = htonl(stats); OLD */
rtcp->r.rr.ssrc = htonl((uint32_t)random()); /* PORT17.5 fix compiler warning */
/* data source being reported */
rtcp->r.rr.rr[0].ssrc = htonl(stats->ssrc);
/* fraction lost since last SR/RR */
if (stats->maxSN-stats->minSN>0)
rtcp->r.rr.rr[0].fraction = (signed)(255*stats->count/(stats->maxSN-stats->minSN));
else
rtcp->r.rr.rr[0].fraction = 0xFF;
/* cumulative number of packets lost (signed!) */
rtcp->r.rr.rr[0].lost = htonl((signed int)(stats->maxSN-stats->minSN-stats->count));
/* extended last sequence number received */
rtcp->r.rr.rr[0].last_seq = htonl(stats->maxSN);
/* interarrival jitter */
rtcp->r.rr.rr[0].jitter = htonl(0xFF);
/* last SR packet from this source */
rtcp->r.rr.rr[0].lsr = htonl(stats->lastTS);
/* delay since last SR packet */
rtcp->r.rr.rr[0].dlsr = htonl(ast_tvdiff_ms(ast_tvnow(),stats->time));
/* Set common headers */
rtcp->common.version = 2;
rtcp->common.p = 0;
rtcp->common.count = 1;
rtcp->common.pt = 201;
/* Length */
rtcp->common.length = htons(7);
}
/* NEW. For SIP */
enum SipMethodsIndex
{
INVITE, /* 0 */
OPTIONS,
ACK,
CANCEL,
BYE,
REFER,
NOTIFY,
MESSAGE,
SUBSCRIBE,
INFO,
MAX_METHODS
};
struct RtspPlayer
{
int fd;
int state;
int cseq;
char* session[2];
int numSessions;
int end; /* Used to exit the main loop */
char* ip; /* destination ip string */
int port; /* destination port */
char* hostport; /* string ip:port */
char* url;
int isIPv6;
char* authorization;
int audioRtp; /* file descriptor */
int audioRtcp; /* file descriptor */
int videoRtp; /* file descriptor */
int videoRtcp; /* file descriptor */
int audioRtpPort; /* source udp port */
int audioRtcpPort; /* source udp port */
int videoRtpPort; /* source udp port */
int videoRtcpPort; /* source udp port */
struct MediaStats audioStats;
struct MediaStats videoStats;
/* NEW. SIP */
char* local_ctrl_ip; /* source IPv4 address string used by SIP */
uint16_t local_ctrl_port; /* source port used by SIP */
int cseqm[MAX_METHODS]; /* SIP differentiates CSeq by sequence number plus Method Sec 20.16 */
int in_a_dialog; /* SIP has a dialog going T/F */
char src_tag[20]; /* SIP random source tag value. Fixed when in a dialog. Hopefully only one dialog per time. */
char peer_tag[20]; /* SIP random tag value received from peer. */
char call_id[100]; /* SIP random call_id value when in a dialog. Hopefully only one call-id per time. */
char branch_id[100];/* SIP random branch_id last transaction. Hopefully only on transaction per time. */
/* SDP */
char session_id[64];/* SDP for SIP sessionID */
};
/* NEW Digest Auth Data */
struct DigestAuthData
{
char nonce[64];
char nc[64];
char cnonce[64];
char qop[16];
char uri[64];
char rx_realm[32];
char opaque[64];
};
/* NEW. For SIP */
static int generateSrcTag(struct RtspPlayer *player)
{
sprintf(player->src_tag,"%08lx", ast_random()); /* Set SIP source Tag */
return 1;
}
/* NEW. For SIP */
static int generateBranch(struct RtspPlayer *player)
{
sprintf(player->branch_id,"z9hG4bKi-%08lx%08lx%08lx%08lx",random(),random(),random(), random() );
return 1;
}
/* NEW. For SIP */
static int generateCallId(struct RtspPlayer *player)
{
if(player->local_ctrl_ip == NULL)
sprintf(player->call_id,"%08lx%08lx%08lx%08lx@foo.bar.com",random(),random(),random(), random() );
else
sprintf(player->call_id,"%08lx%08lx%08lx%08lx@%s",random(),random(),random(), random(),player->local_ctrl_ip );
return 1;
}
/* NEW. For SIP */
static int generateSessionId(struct RtspPlayer *player)
{
sprintf(player->session_id,"158%8ld",random()); /*SDP for SIP RFC3264 Sec 5 requires 64bits; we'll use 32bits */
return 1;
}
static struct RtspPlayer* RtspPlayerCreate(void)
{
/* malloc */
/* struct RtspPlayer* player = (struct RtspPlayer*) malloc(sizeof(struct RtspPlayer)); OLD */
struct RtspPlayer* player = (struct RtspPlayer*) ast_malloc(sizeof(struct RtspPlayer)); /* PORT 17.3 */
/* Initialize */
player->cseq = 1;
player->session[0] = NULL;
player->session[1] = NULL;
player->numSessions = 0;
player->state = RTSP_NONE;
player->end = 0;
player->ip = NULL;
player->hostport = NULL;
player->isIPv6 = 0;
player->port = 0;
player->url = NULL;
player->authorization = NULL;
player->fd = 0; /* Control Protocol (RTSP or SIP) file descriptor */
player->audioRtp = 0; /* file descriptor */
player->audioRtcp = 0; /* file descriptor */
player->videoRtp = 0; /* file descriptor */
player->videoRtcp = 0; /* file descriptor */
player->audioRtpPort = 0; /* Source udp ports */
player->audioRtcpPort = 0; /* Source udp ports */
player->videoRtpPort = 0; /* Source udp ports */
player->videoRtcpPort = 0; /* Source udp ports */
/* ADD. SIP */
player->local_ctrl_ip = NULL; /* source IPv4 address string*/
player->local_ctrl_port = 0; /* source port used by SIP */
int i;
for(i=0;i<MAX_METHODS;i++)
player->cseqm[i] = 1;
player->in_a_dialog = 0; /* SIP has a dialog going T/F */
generateSrcTag(player); /* Set SIP source Tag */
generateBranch(player);
generateCallId(player);
generateSessionId(player);
/* Return */
return player;
}
static void RtspPlayerDestroy(struct RtspPlayer* player)
{
/* free members*/
/* if (player->ip) free(player->ip); OLD */
if (player->ip) ast_free(player->ip);
/* if (player->hostport) free(player->hostport); OLD */
if (player->hostport) ast_free(player->hostport);
/* if (player->url) free(player->url); OLD */
if (player->url) ast_free(player->url);
/* if (player->session[0]) free(player->session[0]); OLD */
if (player->session[0]) ast_free(player->session[0]);
/* if (player->session[1]) free(player->session[1]); OLD */
if (player->session[1]) ast_free(player->session[1]);
/* if (player->authorization) free(player->authorization); OLD */
if (player->authorization) ast_free(player->authorization);
/* ADDED */
if (player->local_ctrl_ip) ast_free(player->local_ctrl_ip);
/* free */
/* free(player); OLD */
ast_free(player);
}
static void RtspPlayerBasicAuthorization(struct RtspPlayer* player,char *username,char *password)
{
char base64[256];
char clear[256];
/* Create authorization header */
/* player->authorization = malloc(128); OLD */
player->authorization = ast_malloc(128); /* PORT 17.3 */
/* Get base 64 from username and password*/
sprintf(clear,"%s:%s",username,password);
/* Encode */
/* ast_base64encode(base64,clear,strlen(clear),256); OLD */
ast_base64encode(base64,(const unsigned char*)clear,strlen(clear),256); /* PORT 17.3. Fix compiler warning */
/* Set heather */
sprintf(player->authorization, "Authorization: Basic %s",base64);
}
/* NEW.
* Use PJSIP to do Digest Authentication
*/
static int RtspPlayerDigestAuthorization(struct RtspPlayer *player,char *cfg_username,\
char *cfg_password, char *cfg_realm, char *nonce, char *nc, \
char *cnonce, char *qop, char *uri, char* rx_realm, char *method, int isSIP)
{
/* PJSIP Notes:
* 1.PJSIP only supports MD5 and AKAv1-MD5.
* Here we assume "algorithm" is "MD5". Others such as SHA-256 are not supported.
* 2.PJSIP appears to support: qop none, or qop=auth, but not qop=auth-int.
* 3.PJSIP requires any char string to have the following structure:
* typedef struct pj_str_t {
* char *ptr;
* pj_size_t slen;
* } pj_str_t;
* These strings are not null terminated, so when used outside of PJSIP, have to add termination.
*/
/* See if Received realm differs from Configured realm */
if (strcmp(cfg_realm,rx_realm)!=0){ /* are not equal */
ast_log(LOG_ERROR,"Received realm %s doesn't match configured realm %s.\n",rx_realm,cfg_realm);
return -1;
}
/* Create authorization header */
player->authorization = ast_malloc(256); /* Freed in RtspPlayerDestroy */
int string_len = 0;
char digest_result[64];
pj_str_t pj_digest_result;
pj_str_t pj_nonce;
pj_str_t pj_nc;
pj_str_t pj_cnonce;
pj_str_t pj_qop;
pj_str_t pj_uri;
pj_str_t pj_rx_realm;
pj_str_t pj_method;
pj_digest_result = pj_str(digest_result);
pj_nonce = pj_str(nonce);
pj_nc = pj_str(nc);
pj_cnonce = pj_str(cnonce);
pj_qop = pj_str(qop);
pj_uri = pj_str(uri);
pj_rx_realm = pj_str(rx_realm);
pj_method = pj_str(method);
pjsip_cred_info cred_info;
cred_info.realm = pj_str(cfg_realm);
cred_info.scheme = pj_str("digest");
cred_info.username = pj_str(cfg_username);
/* cred_info.data_type = PJSIP_CRED_DATA_DIGEST;*/
cred_info.data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
cred_info.data = pj_str(cfg_password);
pjsip_auth_create_digest(&pj_digest_result,&pj_nonce,&pj_nc,\
&pj_cnonce,&pj_qop,&pj_uri,\
&pj_rx_realm, &cred_info, &pj_method);
digest_result[pj_digest_result.slen]='\0'; /*Need to NULL terminate the string */
ast_debug(3,"Digest Result: %s\n",digest_result);
if(isSIP){
/*
* RFC 3261 p226: dig-resp: username="string",realm="string",
* nonce="string",uri="string",response="32LHex",
* no one quotes MD5 algorithm=<"MD5"|token>,cnonce="string",opaque="string",
* no one quotes qopvalue qop="qop-value",nc=8Lhex,token=<token|"string'>
*/
/* Auth Header should look something like the following:
* Note: comma-separated list (space after comma) ex. sect 20.44, 22.2
* Authorization: Digest\r\n username="my_name", realm="streaming_server",
* nonce="a50392b361ce351d9a95e73a45a6b133", uri="sip:0002D151D42F@192.168.0.43:5060",
* response="b8325b600081488dde51435dda7e3162", algorithm=MD5\r\n
*/
string_len = sprintf(player->authorization,
"Authorization: Digest "
"username=\"%s\", realm=\"%s\", nonce=\"%s\", uri=\"%s\", response=\"%s\", algorithm=MD5",
cfg_username,rx_realm,nonce,uri,digest_result);
if(cnonce != NULL)
string_len += sprintf(player->authorization+string_len,",cnonce\"%s\"",cnonce);
if(qop != NULL)
string_len += sprintf(player->authorization+string_len,",qop\"%s\"",qop);
if(nc != NULL)
string_len += sprintf(player->authorization+string_len,",nc\"%s\"",nc);
}
else{ /* RTSP NOT BEEN TESTED FOR DIGEST AUTH */
sprintf(player->authorization, "Authorization: Digest %s",pj_digest_result.ptr);
ast_log(LOG_WARNING,"RTSP not tested for Digest Authentication.\n");
}
ast_debug(3,"Auth: \n%s\n",player->authorization);
return 1;
}
/*
* COMMENT:
* GetUdpPorts() forces the source ports for RTP/RTCP to be paired odd/even respectively.
*/
static void GetUdpPorts(int *a,int *b,int *p,int *q,int isIPv6)
{
struct sockaddr *sendAddr;
int size;
int len;
int PF;
unsigned short *port;
/* If it is ipv6 */
if (isIPv6)
{
/* Set size*/
size = sizeof(struct sockaddr_in6);
/*Create address */
/* sendAddr = (struct sockaddr *)malloc(size); OLD */
sendAddr = (struct sockaddr *)ast_malloc(size);
/* empty addres */
memset(sendAddr,0,size);
/*Set family */
((struct sockaddr_in6*)sendAddr)->sin6_family = AF_INET6;
/* Set PF */
PF = PF_INET6;
/* Get port */
port = &(((struct sockaddr_in6 *)sendAddr)->sin6_port);
} else {
/* Set size*/
size = sizeof(struct sockaddr_in);
/*Create address */
/* sendAddr = (struct sockaddr *)malloc(size); OLD */
sendAddr = (struct sockaddr *)ast_malloc(size);
/* empty addres */
memset(sendAddr,0,size);
/*Set family */
((struct sockaddr_in *)sendAddr)->sin_family = AF_INET;
/* Set PF */
PF = PF_INET;
/* Get port */
port = &(((struct sockaddr_in *)sendAddr)->sin_port);
}
/* Create sockets */
*a = socket(PF,SOCK_DGRAM,0);
bind(*a,sendAddr,size);
*b = socket(PF,SOCK_DGRAM,0);
bind(*b,sendAddr,size);
/* Get socket ports */
len = size;
/* getsockname(*a,sendAddr,&len); OLD */
getsockname(*a,sendAddr,(socklen_t*)&len); /* PORT17.3. fix compiler warning */
*p = ntohs(*port);
len = size;
/* getsockname(*b,sendAddr,&len); OLD */
getsockname(*b,sendAddr,(socklen_t*)&len); /* PORT17.3. fix compiler warning */
*q = ntohs(*port);
/* ast_log(LOG_DEBUG,"-GetUdpPorts [%d,%d]\n",*p,*q); OLD */
ast_debug(4,"-GetUdpPorts initial [%d,%d]\n",*p,*q);
/* Create audio sockets */
while ( *p%2 || *p+1!=*q )
{
/* Close first */
close(*a);
/* Move one forward */
*a = *b;
*p = *q;
/* Create new socket */
*b = socket(PF,SOCK_DGRAM,0);
/* Get port */
if (*p>0)
*port = htons(*p+1);
else
*port = htons(0);
bind(*b,sendAddr,size);
len = size;
/* getsockname(*b,sendAddr,&len); OLD */
getsockname(*b,sendAddr,(socklen_t*)&len); /* PORT17.3. fix complier warning */
*q = ntohs(*port);
/* ast_log(LOG_DEBUG,"-GetUdpPorts [%d,%d]\n",*p,*q); OLD */
ast_debug(4,"-GetUdpPorts loop [%d,%d]\n",*p,*q);
}
ast_debug(3,"-GetUdpPorts final [%d,%d]\n",*p,*q); /* ADDED */
/* Free Address*/
/* free(sendAddr); OLD */
ast_free(sendAddr);
}
static void SetNonBlocking(int fd)
{
/* Get flags */
int flags = fcntl(fd,F_GETFD);
/* Set socket non-blocking */
fcntl(fd,F_SETFD,flags | O_NONBLOCK);
}
/*
* COMMENT:
* The following only sets up the socket structure for destination ip address and port.
* It does not get the ip address.
*/
static struct sockaddr* GetIPAddr(const char *ip, int port,int isIPv6,int *size,int *PF)
{
struct sockaddr * sendAddr;
/* If it is ipv6 */
if (isIPv6)
{
/* Set size*/
*size = sizeof(struct sockaddr_in6);
/*Create address */
/* sendAddr = (struct sockaddr *)malloc(*size); OLD */
sendAddr = (struct sockaddr *)ast_malloc(*size);
/* empty addres */
memset(sendAddr,0,*size);
/* Set PF */
*PF = PF_INET6;
/*Set family */
((struct sockaddr_in6 *)sendAddr)->sin6_family = AF_INET6;
/* Set Address */
inet_pton(AF_INET6,ip,&((struct sockaddr_in6*)sendAddr)->sin6_addr);
/* Set port */
((struct sockaddr_in6 *)sendAddr)->sin6_port = htons(port);
} else {
/* Set size*/
*size = sizeof(struct sockaddr_in);
/*Create address */
/* sendAddr = (struct sockaddr *)malloc(*size); OLD */
sendAddr = (struct sockaddr *)ast_malloc(*size);
/* empty addres */
memset(sendAddr,0,*size);
/*Set family */
((struct sockaddr_in*)sendAddr)->sin_family = AF_INET;
/* Set PF */
*PF = PF_INET;
/* Set Address */
((struct sockaddr_in*)sendAddr)->sin_addr.s_addr = inet_addr(ip);
/* Set port */
((struct sockaddr_in *)sendAddr)->sin_port = htons(port);
}
return sendAddr;
}
static int RtspPlayerConnect(struct RtspPlayer *player, const char *ip, int port,int isIPv6, int isUDP) /*ADDED isUDP */
{
struct sockaddr * sendAddr;
int size;
int PF;
/* ADDED */
char local_ip[100];
uint16_t local_port;
struct sockaddr_in name;
socklen_t namelen = sizeof(name);
int err;
/* Setup struct for Dest address port */
sendAddr = GetIPAddr(ip,port,isIPv6,&size,&PF);
/* open Control socket */
if(isUDP) /* ADDED. SIP uses UDP. RTSP uses TCP*/
player->fd = socket(PF,SOCK_DGRAM,0);
else
player->fd = socket(PF,SOCK_STREAM,0);
/* Create/Open audio datagram sockets and ports for RTP and RTCP*/
GetUdpPorts(&player->audioRtp,&player->audioRtcp,&player->audioRtpPort,&player->audioRtcpPort,isIPv6);
/* Create/Open video datagram sockets and ports for RTP and RTCP*/
GetUdpPorts(&player->videoRtp,&player->videoRtcp,&player->videoRtpPort,&player->videoRtcpPort,isIPv6);
/* Set non blocking */
SetNonBlocking(player->fd);
SetNonBlocking(player->audioRtp);
SetNonBlocking(player->audioRtcp);
SetNonBlocking(player->videoRtp);
SetNonBlocking(player->videoRtcp);
/* Connect */
if (connect(player->fd,sendAddr,size)<0)
{
/* Free mem */
/* free(sendAddr); OLD */
ast_free(sendAddr);
/* Exit */
return 0;
}
/* ADDED. Get local IP and source Port in text format for Control Protocol*/
err = getsockname(player->fd, (struct sockaddr*) &name, &namelen);
if(err !=0)
ast_log(LOG_ERROR,"Could not get local IP address\n");
const char* p = inet_ntop(AF_INET, &name.sin_addr, local_ip, 100);
if(p ==NULL)
ast_log(LOG_ERROR,"Could not convert local IP address\n");
local_port = ntohs(name.sin_port);
player->local_ctrl_ip = ast_strdup(local_ip);
player->local_ctrl_port = local_port;
ast_debug(3,"Local Ctrl IP: %s, Port: %i\n",player->local_ctrl_ip,player->local_ctrl_port);
/* Set ip v6 */
player->isIPv6 = isIPv6;
/* copy ip & port*/
/* player->ip = strdup(ip); OLD */
player->ip = ast_strdup(ip);
player->port = port;
/* create hostport */
/* player->hostport = (char*)malloc(strlen(ip)+8); OLD */
player->hostport = (char*)ast_malloc(strlen(ip)+8);
/* If it is ipv6 */
if (isIPv6)
/* In brackets */
sprintf(player->hostport,"[%s]",player->ip);
else
/* normal*/
strcpy(player->hostport,player->ip);
/* If not standard port */
if (port!=554)
{
/* Append port to ip */
/* PORT 17.5, fix compiler warning about sprintf arg1 alias w. arg3 */
/* sprintf(player->hostport,"%s:%d",player->hostport,player->port); OLD */
char temp[100];
strcpy(temp,player->hostport);
sprintf(player->hostport,"%s:%d",temp,player->port);
}
/* Free mem */
/* free(sendAddr); OLD */
ast_free(sendAddr);
/* conected */
return 1;
}
static int RtspPlayerAddSession(struct RtspPlayer *player,char *session)
{
int i;
char *p;
/* If max sessions reached */
if (player->numSessions == 2)
/* Exit */
return 0;
/* Check if it has parameters */
if ((p=strchr(session,';'))>0)
/* Remove then */
*p = 0;
/* Check if we have that session already */
for (i=0;i<player->numSessions;i++)