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	When dealing with a lot of video streams on WebRTC the resulting SDPs can grow to be quite large. This effectively doubles the maximum size to allow more streams to exist. The res_http_websocket module has also been changed to use a buffer on the session for reading in packets to ensure that the stack space usage is not excessive. Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01
		
			
				
	
	
		
			90 lines
		
	
	
		
			2.6 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			90 lines
		
	
	
		
			2.6 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * Asterisk config_site.h
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|  */
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| 
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| #include <sys/select.h>
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| 
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| /*
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|  * Since both pjproject and asterisk source files will include config_site.h,
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|  * we need to make sure that only pjproject source files include asterisk_malloc_debug.h.
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|  */
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| #if !defined(_ASTERISK_ASTMM_H)
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| #include "asterisk_malloc_debug.h"
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| #endif
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| 
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| /*
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|  * Defining PJMEDIA_HAS_SRTP to 0 does NOT disable Asterisk's ability to use srtp.
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|  * It only disables the pjmedia srtp transport which Asterisk doesn't use.
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|  * The reason for the disable is that while Asterisk works fine with older libsrtp
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|  * versions, newer versions of pjproject won't compile with them.
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|  */
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| #define PJMEDIA_HAS_SRTP 0
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| 
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| #define PJ_HAS_IPV6 1
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| #if !defined(AST_DEVMODE) && !defined(PJPROJECT_BUNDLED_ASSERTIONS)
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| #define NDEBUG 1
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| #endif
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| 
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| #define PJ_MAX_HOSTNAME (256)
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| #define PJSIP_MAX_URL_SIZE (512)
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| #ifdef PJ_HAS_LINUX_EPOLL
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| #define PJ_IOQUEUE_MAX_HANDLES	(5000)
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| #else
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| #define PJ_IOQUEUE_MAX_HANDLES	(FD_SETSIZE)
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| #endif
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| #define PJ_IOQUEUE_HAS_SAFE_UNREG 1
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| #define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)
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| 
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| #define PJ_SCANNER_USE_BITWISE	0
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| #define PJ_OS_HAS_CHECK_STACK	0
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| 
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| #ifndef PJ_LOG_MAX_LEVEL
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| #define PJ_LOG_MAX_LEVEL		6
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| #endif
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| 
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| #define PJ_ENABLE_EXTRA_CHECK	1
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| #define PJSIP_MAX_TSX_COUNT		((64*1024)-1)
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| #define PJSIP_MAX_DIALOG_COUNT	((64*1024)-1)
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| #define PJSIP_UDP_SO_SNDBUF_SIZE	(512*1024)
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| #define PJSIP_UDP_SO_RCVBUF_SIZE	(512*1024)
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| #define PJ_DEBUG			0
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| #define PJSIP_SAFE_MODULE		0
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| #define PJ_HAS_STRICMP_ALNUM		0
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| 
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| /*
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|  * Do not ever enable PJ_HASH_USE_OWN_TOLOWER because the algorithm is
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|  * inconsistently used when calculating the hash value and doesn't
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|  * convert the same characters as pj_tolower()/tolower().  Thus you
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|  * can get different hash values if the string hashed has certain
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|  * characters in it.  (ASCII '@', '[', '\\', ']', '^', and '_')
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|  */
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| #undef PJ_HASH_USE_OWN_TOLOWER
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| 
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| /*
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|   It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
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|   Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
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| */
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| #undef PJSIP_UNESCAPE_IN_PLACE
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| #define PJSIP_MAX_PKT_LEN			65535
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| 
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| #undef PJ_TODO
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| #define PJ_TODO(x)
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| 
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| /* Defaults too low for WebRTC */
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| #define PJ_ICE_MAX_CAND 64
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| #define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
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| 
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| /* Increase limits to allow more formats */
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| #define	PJMEDIA_MAX_SDP_FMT   64
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| #define	PJMEDIA_MAX_SDP_BANDW   4
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| #define	PJMEDIA_MAX_SDP_ATTR   (PJMEDIA_MAX_SDP_FMT*2 + 4)
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| #define	PJMEDIA_MAX_SDP_MEDIA   16
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| 
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| /*
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|  * Turn off the periodic sending of CRLNCRLN.  Default is on (90 seconds),
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|  * which conflicts with the global section's keep_alive_interval option in
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|  * pjsip.conf.
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|  */
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| #define PJSIP_TCP_KEEP_ALIVE_INTERVAL	0
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| #define PJSIP_TLS_KEEP_ALIVE_INTERVAL	0
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