by the asterisk project. This allows WebRTC to work correctly in asterisk out of the box [1] - Also import some patches to pjsip from the asterisk project. These patches have already been integrated in upstream pjsip development sources [2] Reported by: zhenya1993lzn@yandex.ru [1] Obtained from: https://github.com/asterisk/asterisk/tree/16.3/third-party/pjproject/patches [2] MFH: 2019Q2
40 lines
1.1 KiB
C
40 lines
1.1 KiB
C
/*
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* Custom options imported from asterisk provided customizations.
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*
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* $FreeBSD$
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*/
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#include <sys/select.h>
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#define PJ_MAX_HOSTNAME (256)
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#define PJSIP_MAX_URL_SIZE (512)
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#define PJ_IOQUEUE_MAX_HANDLES (FD_SETSIZE)
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#define PJ_SCANNER_USE_BITWISE 0
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#define PJ_LOG_MAX_LEVEL 6
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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 PJSIP_MAX_PKT_LEN 32000
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/* Defaults too low for WebRTC */
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#define PJ_ICE_MAX_CAND 32
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#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
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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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* 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 in asterisk.
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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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