mirror of
https://github.com/asterisk/asterisk.git
synced 2025-09-03 11:25:35 +00:00
edd7ae85e8
The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how many pairs of local/remote candidates will be made. If for some reason we reach this upper bound, ICE will generally fail and no media will flow between the browser and Asterisk. This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of pairs of candidates we'd theoretically allow, which is PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame Docker), this is far too low to allow WebRTC calls to succeed. Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed even when the system Asterisk was running on had quite a few virtual interfaces. Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55
66 lines
2.0 KiB
C
66 lines
2.0 KiB
C
/*
|
|
* Asterisk config_site.h
|
|
*/
|
|
|
|
#include <sys/select.h>
|
|
|
|
/*
|
|
* Since both pjproject and asterisk source files will include config_site.h,
|
|
* we need to make sure that only pjproject source files include asterisk_malloc_debug.h.
|
|
*/
|
|
#if defined(MALLOC_DEBUG) && !defined(_ASTERISK_ASTMM_H)
|
|
#include "asterisk_malloc_debug.h"
|
|
#endif
|
|
|
|
/*
|
|
* Defining PJMEDIA_HAS_SRTP to 0 does NOT disable Asterisk's ability to use srtp.
|
|
* It only disables the pjmedia srtp transport which Asterisk doesn't use.
|
|
* The reason for the disable is that while Asterisk works fine with older libsrtp
|
|
* versions, newer versions of pjproject won't compile with them.
|
|
*/
|
|
#define PJMEDIA_HAS_SRTP 0
|
|
|
|
#define PJ_HAS_IPV6 1
|
|
#define NDEBUG 1
|
|
#define PJ_MAX_HOSTNAME (256)
|
|
#define PJSIP_MAX_URL_SIZE (512)
|
|
#ifdef PJ_HAS_LINUX_EPOLL
|
|
#define PJ_IOQUEUE_MAX_HANDLES (5000)
|
|
#else
|
|
#define PJ_IOQUEUE_MAX_HANDLES (FD_SETSIZE)
|
|
#endif
|
|
#define PJ_IOQUEUE_HAS_SAFE_UNREG 1
|
|
#define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)
|
|
|
|
#define PJ_SCANNER_USE_BITWISE 0
|
|
#define PJ_OS_HAS_CHECK_STACK 0
|
|
#define PJ_LOG_MAX_LEVEL 3
|
|
#define PJ_ENABLE_EXTRA_CHECK 1
|
|
#define PJSIP_MAX_TSX_COUNT ((64*1024)-1)
|
|
#define PJSIP_MAX_DIALOG_COUNT ((64*1024)-1)
|
|
#define PJSIP_UDP_SO_SNDBUF_SIZE (512*1024)
|
|
#define PJSIP_UDP_SO_RCVBUF_SIZE (512*1024)
|
|
#define PJ_DEBUG 0
|
|
#define PJSIP_SAFE_MODULE 0
|
|
#define PJ_HAS_STRICMP_ALNUM 0
|
|
#define PJ_HASH_USE_OWN_TOLOWER 1
|
|
/*
|
|
It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
|
|
Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
|
|
*/
|
|
#undef PJSIP_UNESCAPE_IN_PLACE
|
|
#define PJSIP_MAX_PKT_LEN 6000
|
|
|
|
#undef PJ_TODO
|
|
#define PJ_TODO(x)
|
|
|
|
/* Defaults too low for WebRTC */
|
|
#define PJ_ICE_MAX_CAND 32
|
|
#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
|
|
|
|
/* Increase limits to allow more formats */
|
|
#define PJMEDIA_MAX_SDP_FMT 64
|
|
#define PJMEDIA_MAX_SDP_BANDW 4
|
|
#define PJMEDIA_MAX_SDP_ATTR (PJMEDIA_MAX_SDP_FMT*2 + 4)
|
|
#define PJMEDIA_MAX_SDP_MEDIA 16
|