diff options
author | Olle Johansson <oej@edvina.net> | 2006-03-27 03:35:49 +0000 |
---|---|---|
committer | Olle Johansson <oej@edvina.net> | 2006-03-27 03:35:49 +0000 |
commit | 83d9331261fdb0a04bf577c1b822842d9b68d743 (patch) | |
tree | 123ad8fafd10ea5f8fe77291b37c6a7ebe28a0a3 /channels | |
parent | 18de2b7787b216f0d90cb49c1e7e8c490a7ca2ea (diff) |
Issue #5427
- Enable videosupport per device
- Implement maxcallbitrate setting for video calls
Patch by John Martin, thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r-- | channels/chan_sip.c | 102 |
1 files changed, 85 insertions, 17 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 08a529018..5b94d510a 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -362,7 +362,6 @@ static const struct cfsip_options { #define DEFAULT_NOTIFYMIME "application/simple-message-summary" #define DEFAULT_MWITIME 10 #define DEFAULT_ALLOWGUEST TRUE -#define DEFAULT_VIDEOSUPPORT FALSE #define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */ #define DEFAULT_COMPACTHEADERS FALSE #define DEFAULT_TOS FALSE @@ -373,10 +372,12 @@ static const struct cfsip_options { #define DEFAULT_AUTOCREATEPEER FALSE #define DEFAULT_QUALIFY FALSE #define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */ +#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */ #ifndef DEFAULT_USERAGENT #define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */ #endif + /* Default setttings are used as a channel setting and as a default when configuring devices */ static char default_context[AST_MAX_CONTEXT]; @@ -388,6 +389,7 @@ static char default_notifymime[AST_MAX_EXTENSION]; static int default_qualify; /*!< Default Qualify= setting */ static char default_vmexten[AST_MAX_EXTENSION]; static char default_musicclass[MAX_MUSICCLASS]; /*!< Global music on hold class */ +static int default_maxcallbitrate; /*!< Maximum bitrate for call */ static struct ast_codec_pref default_prefs; /*!< Default codec prefs */ /* Global settings only apply to the channel */ @@ -407,7 +409,6 @@ static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, thi the global setting is in globals_flag_page2 */ static int global_mwitime; /*!< Time between MWI checks for peers */ static int global_tos; /*!< IP Type of service */ -static int global_videosupport; /*!< Videosupport on or off */ static int compactheaders; /*!< send compact sip headers */ static int recordhistory; /*!< Record SIP history. Off by default */ static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */ @@ -609,11 +610,13 @@ struct sip_auth { #define SIP_PAGE2_DEBUG_CONSOLE (1 << 6) #define SIP_PAGE2_DYNAMIC (1 << 7) /*!< Dynamic Peers register with Asterisk */ #define SIP_PAGE2_SELFDESTRUCT (1 << 8) /*!< Automatic peers need to destruct themselves */ +#define SIP_PAGE2_VIDEOSUPPORT (1 << 9) #define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 10) /*!< Allow subscriptions from this peer? */ #define SIP_PAGE2_ALLOWOVERLAP (1 << 11) /*!< Allow overlap dialing ? */ + #define SIP_PAGE2_FLAGS_TO_COPY \ - (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP) + (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT) /* SIP packet flags */ #define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */ @@ -682,6 +685,7 @@ static struct sip_pvt { int peercapability; /*!< Supported peer capability */ int prefcodec; /*!< Preferred codec (outbound only) */ int noncodeccapability; + int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */ int callingpres; /*!< Calling presentation */ int authtries; /*!< Times we've tried to authenticate */ int expiry; /*!< How long we take to expire */ @@ -782,6 +786,7 @@ struct sip_user { int call_limit; /*!< Limit of concurrent calls */ struct ast_ha *ha; /*!< ACL setting */ struct ast_variable *chanvars; /*!< Variables to set for channel created by user */ + int maxcallbitrate; /*!< Maximum Bitrate for a video call */ }; /*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */ @@ -826,7 +831,8 @@ struct sip_peer { ast_group_t pickupgroup; /*!< Pickup group */ struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */ struct sockaddr_in addr; /*!< IP address of peer */ - + int maxcallbitrate; /*!< Maximum Bitrate for a video call */ + /* Qualification */ struct sip_pvt *call; /*!< Call pointer */ int pokeexpire; /*!< When to expire poke (qualify= checking) */ @@ -1926,7 +1932,12 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer) } ast_copy_flags(r, peer, SIP_FLAGS_TO_COPY); + ast_copy_flags((&r->flags_page2),(&peer->flags_page2), SIP_PAGE2_FLAGS_TO_COPY); r->capability = peer->capability; + if (!ast_test_flag((&r->flags_page2), SIP_PAGE2_VIDEOSUPPORT) && r->vrtp) { + ast_rtp_destroy(r->vrtp); + r->vrtp = NULL; + } r->prefs = peer->prefs; if (r->rtp) { if (option_debug) @@ -1985,7 +1996,8 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer) r->rtpkeepalive = peer->rtpkeepalive; if (peer->call_limit) ast_set_flag(r, SIP_CALL_LIMIT); - + r->maxcallbitrate = peer->maxcallbitrate; + return 0; } @@ -2852,6 +2864,7 @@ static int sip_indicate(struct ast_channel *ast, int condition) case AST_CONTROL_VIDUPDATE: /* Request a video frame update */ if (p->vrtp && !ast_test_flag(p, SIP_NOVIDEO)) { transmit_info_with_vidupdate(p); + /* ast_rtcp_send_h261fur(p->vrtp); */ res = 0; } else res = -1; @@ -3230,6 +3243,9 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si } else { memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); } + + ast_copy_flags(p, &global_flags, SIP_FLAGS_TO_COPY); + ast_copy_flags((&p->flags_page2),(&global_flags_page2), SIP_PAGE2_FLAGS_TO_COPY); p->branch = thread_safe_rand(); make_our_tag(p->tag, sizeof(p->tag)); @@ -3238,10 +3254,10 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si if (sip_methods[intended_method].need_rtp) { p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); - if (global_videosupport) + if (ast_test_flag((&p->flags_page2), SIP_PAGE2_VIDEOSUPPORT)) p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr); - if (!p->rtp || (global_videosupport && !p->vrtp)) { - ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", global_videosupport ? "and video" : "", strerror(errno)); + if (!p->rtp || (ast_test_flag((&p->flags_page2), SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) { + ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", ast_test_flag((&p->flags_page2), SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno)); ast_mutex_destroy(&p->lock); if (p->chanvars) { ast_variables_destroy(p->chanvars); @@ -3256,6 +3272,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si p->rtptimeout = global_rtptimeout; p->rtpholdtimeout = global_rtpholdtimeout; p->rtpkeepalive = global_rtpkeepalive; + p->maxcallbitrate = default_maxcallbitrate; } if (useglobal_nat && sin) { @@ -3275,7 +3292,6 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si build_callid_pvt(p); else ast_string_field_set(p, callid, callid); - ast_copy_flags(p, &global_flags, SIP_FLAGS_TO_COPY); /* Assign default music on hold class */ ast_string_field_set(p, musicclass, default_musicclass); p->capability = global_capability; @@ -4501,6 +4517,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) char o[256]; char c[256]; char t[256]; + char b[256]; char m_audio[256]; char m_video[256]; char a_audio[1024]; @@ -4571,6 +4588,8 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr)); snprintf(s, sizeof(s), "s=session\r\n"); snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr)); + if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */ + snprintf(b, sizeof(b), "b=CT:%d\r\n", p->maxcallbitrate); snprintf(t, sizeof(t), "t=0 0\r\n"); ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port)); @@ -4616,7 +4635,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) } /* Now send any other common codecs, and non-codec formats: */ - for (x = 1; x <= ((global_videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) { + for (x = 1; x <= ((ast_test_flag((&p->flags_page2), SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) { if (!(capability & x)) continue; @@ -4655,7 +4674,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio); if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */ - len += strlen(m_video) + strlen(a_video); + len += strlen(m_video) + strlen(a_video) + strlen(b); add_header(resp, "Content-Type", "application/sdp"); add_header_contentLength(resp, len); @@ -4663,6 +4682,8 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) add_line(resp, o); add_line(resp, s); add_line(resp, c); + if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */ + add_line(resp, b); add_line(resp, t); add_line(resp, m_audio); add_line(resp, a_audio); @@ -7200,6 +7221,7 @@ static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipme if (!(res = check_auth(p, req, user->name, user->secret, user->md5secret, sipmethod, uri, reliable, ignore))) { sip_cancel_destroy(p); ast_copy_flags(p, user, SIP_FLAGS_TO_COPY); + ast_copy_flags((&p->flags_page2),(&user->flags_page2), SIP_PAGE2_FLAGS_TO_COPY); /* Copy SIP extensions profile from INVITE */ if (p->sipoptions) user->sipoptions = p->sipoptions; @@ -7233,6 +7255,11 @@ static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipme p->callingpres = user->callingpres; p->capability = user->capability; p->jointcapability = user->capability; + p->maxcallbitrate = user->maxcallbitrate; + if (!ast_test_flag((&p->flags_page2), SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) { + ast_rtp_destroy(p->vrtp); + p->vrtp = NULL; + } if (p->peercapability) p->jointcapability &= p->peercapability; if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO)) @@ -7352,6 +7379,11 @@ static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipme p->jointcapability = peer->capability; if (p->peercapability) p->jointcapability &= p->peercapability; + p->maxcallbitrate = peer->maxcallbitrate; + if (!ast_test_flag((&p->flags_page2), SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) { + ast_rtp_destroy(p->vrtp); + p->vrtp = NULL; + } if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO)) p->noncodeccapability |= AST_RTP_DTMF; else @@ -7730,6 +7762,7 @@ static int _sip_show_peers(int fd, int *total, struct mansession *s, struct mess "IPport: %d\r\n" "Dynamic: %s\r\n" "Natsupport: %s\r\n" + "Video Support: %s\r\n" "ACL: %s\r\n" "Status: %s\r\n" "RealtimeDevice: %s\r\n\r\n", @@ -7739,7 +7772,8 @@ static int _sip_show_peers(int fd, int *total, struct mansession *s, struct mess ntohs(iterator->addr.sin_port), ast_test_flag((&iterator->flags_page2), SIP_PAGE2_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */ (ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */ - iterator->ha ? "yes" : "no", /* iterator/deny */ + ast_test_flag((&iterator->flags_page2), SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */ + iterator->ha ? "yes" : "no", /* permit/deny */ status, realtimepeers ? (ast_test_flag(iterator, SIP_REALTIME) ? "yes":"no") : "no"); } @@ -8118,6 +8152,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message ast_cli(fd, " Call limit : %d\n", peer->call_limit); ast_cli(fd, " Dynamic : %s\n", (ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC)?"Yes":"No")); ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>")); + ast_cli(fd, " MaxCallBR : %dkbps\n", peer->maxcallbitrate); ast_cli(fd, " Expire : %d\n", peer->expire); ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(peer, SIP_INSECURE_PORT), ast_test_flag(peer, SIP_INSECURE_INVITE))); ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(peer, SIP_NAT))); @@ -8125,6 +8160,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message ast_cli(fd, " CanReinvite : %s\n", (ast_test_flag(peer, SIP_CAN_REINVITE)?"Yes":"No")); ast_cli(fd, " PromiscRedir : %s\n", (ast_test_flag(peer, SIP_PROMISCREDIR)?"Yes":"No")); ast_cli(fd, " User=Phone : %s\n", (ast_test_flag(peer, SIP_USEREQPHONE)?"Yes":"No")); + ast_cli(fd, " Video Support: %s\n", (ast_test_flag((&peer->flags_page2), SIP_PAGE2_VIDEOSUPPORT)?"Yes":"No")); ast_cli(fd, " Trust RPID : %s\n", (ast_test_flag(peer, SIP_TRUSTRPID) ? "Yes" : "No")); ast_cli(fd, " Send RPID : %s\n", (ast_test_flag(peer, SIP_SENDRPID) ? "Yes" : "No")); ast_cli(fd, " Subscriptions: %s\n", ast_test_flag(&peer->flags_page2, SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No"); @@ -8196,6 +8232,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message astman_append(s, "VoiceMailbox: %s\r\n", peer->mailbox); astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent); astman_append(s, "Call limit: %d\r\n", peer->call_limit); + astman_append(s, "MaxCallBR: %dkbps\r\n", peer->maxcallbitrate); astman_append(s, "Dynamic: %s\r\n", (ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC)?"Y":"N")); astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "")); astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched,peer->expire)); @@ -8205,6 +8242,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(peer, SIP_CAN_REINVITE)?"Y":"N")); astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(peer, SIP_PROMISCREDIR)?"Y":"N")); astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(peer, SIP_USEREQPHONE)?"Y":"N")); + astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag((&peer->flags_page2), SIP_PAGE2_VIDEOSUPPORT)?"Y":"N")); /* - is enumerated */ astman_append(s, "SIP-DTMFmode %s\r\n", dtmfmode2str(ast_test_flag(peer, SIP_DTMF))); @@ -8353,7 +8391,7 @@ static int sip_show_settings(int fd, int argc, char *argv[]) ast_cli(fd, "----------------\n"); ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port)); ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(tmp, sizeof(tmp), bindaddr.sin_addr)); - ast_cli(fd, " Videosupport: %s\n", global_videosupport ? "Yes" : "No"); + ast_cli(fd, " Videosupport: %s\n", ast_test_flag((&global_flags_page2), SIP_PAGE2_VIDEOSUPPORT) ? "Yes" : "No"); ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No"); ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No"); ast_cli(fd, " Allow subscriptions: %s\n", ast_test_flag(&global_flags_page2, SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No"); @@ -8401,6 +8439,7 @@ static int sip_show_settings(int fd, int argc, char *argv[]) ast_cli(fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout); ast_cli(fd, " Outbound reg. attempts: %d\n", global_regattempts_max); ast_cli(fd, " Notify ringing state: %s\n", global_notifyringing ? "Yes" : "No"); + ast_cli(fd, " Max Call Bitrate: %dkbps\r\n", default_maxcallbitrate); ast_cli(fd, "\nDefault Settings:\n"); ast_cli(fd, "-----------------\n"); ast_cli(fd, " Context: %s\n", default_context); @@ -12144,7 +12183,18 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int user->callingpres = ast_parse_caller_presentation(v->value); if (user->callingpres == -1) user->callingpres = atoi(v->value); - } + } else if (!strcasecmp(v->name, "maxcallbitrate")) { + user->maxcallbitrate = atoi(v->value); + if (user->maxcallbitrate < 0) + user->maxcallbitrate = default_maxcallbitrate; + } else if (!strcasecmp(v->name, "videosupport")) { + if (ast_test_flag((&global_flags_page2), SIP_PAGE2_VIDEOSUPPORT)) { + if (ast_true(v->value)) + ast_set_flag((&user->flags_page2), SIP_PAGE2_VIDEOSUPPORT); + else + ast_clear_flag((&user->flags_page2), SIP_PAGE2_VIDEOSUPPORT); + } + } } ast_copy_flags(user, &userflags, mask.flags); ast_free_ha(oldha); @@ -12162,6 +12212,7 @@ static void set_peer_defaults(struct sip_peer *peer) peer->pokeexpire = -1; peer->addr.sin_port = htons(DEFAULT_SIP_PORT); } + ast_copy_flags((&peer->flags_page2), &global_flags_page2, SIP_PAGE2_FLAGS_TO_COPY); ast_copy_flags(peer, &global_flags, SIP_FLAGS_TO_COPY); ast_copy_flags(&peer->flags_page2, &global_flags_page2, SIP_PAGE2_FLAGS_TO_COPY); strcpy(peer->context, default_context); @@ -12171,6 +12222,7 @@ static void set_peer_defaults(struct sip_peer *peer) peer->addr.sin_family = AF_INET; peer->defaddr.sin_family = AF_INET; peer->capability = global_capability; + peer->maxcallbitrate = default_maxcallbitrate; peer->rtptimeout = global_rtptimeout; peer->rtpholdtimeout = global_rtpholdtimeout; peer->rtpkeepalive = global_rtpkeepalive; @@ -12302,7 +12354,14 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int ast_copy_string(peer->fromdomain, v->value, sizeof(peer->fromdomain)); else if (!strcasecmp(v->name, "usereqphone")) ast_set2_flag(peer, ast_true(v->value), SIP_USEREQPHONE); - else if (!strcasecmp(v->name, "fromuser")) + else if (!strcasecmp(v->name, "videosupport")) { + if (ast_test_flag((&global_flags_page2), SIP_PAGE2_VIDEOSUPPORT)) { + if (ast_true(v->value)) + ast_set_flag((&peer->flags_page2), SIP_PAGE2_VIDEOSUPPORT); + else + ast_clear_flag((&peer->flags_page2), SIP_PAGE2_VIDEOSUPPORT); + } + } else if (!strcasecmp(v->name, "fromuser")) ast_copy_string(peer->fromuser, v->value, sizeof(peer->fromuser)); else if (!strcasecmp(v->name, "host") || !strcasecmp(v->name, "outboundproxy")) { if (!strcasecmp(v->value, "dynamic")) { @@ -12426,6 +12485,10 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno); peer->maxms = 0; } + } else if (!strcasecmp(v->name, "maxcallbitrate")) { + peer->maxcallbitrate = atoi(v->value); + if (peer->maxcallbitrate < 0) + peer->maxcallbitrate = default_maxcallbitrate; } } if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC) && realtime) { @@ -12510,7 +12573,6 @@ static int reload_config(enum channelreloadreason reason) ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime)); ast_copy_string(global_realm, DEFAULT_REALM, sizeof(global_realm)); ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid)); - global_videosupport = DEFAULT_VIDEOSUPPORT; compactheaders = DEFAULT_COMPACTHEADERS; global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT; global_regattempts_max = 0; @@ -12534,6 +12596,7 @@ static int reload_config(enum channelreloadreason reason) default_language[0] = '\0'; default_fromdomain[0] = '\0'; default_qualify = DEFAULT_QUALIFY; + default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE; ast_copy_string(default_musicclass, DEFAULT_MUSICCLASS, sizeof(default_musicclass)); ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten)); ast_set_flag(&global_flags, SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */ @@ -12549,6 +12612,7 @@ static int reload_config(enum channelreloadreason reason) global_relaxdtmf = FALSE; global_callevents = FALSE; global_t1min = DEFAULT_T1MIN; + ast_clear_flag(&global_flags_page2, SIP_PAGE2_VIDEOSUPPORT); /* Read the [general] config section of sip.conf (or from realtime config) */ for (v = ast_variable_browse(cfg, "general"); v; v = v->next) { @@ -12605,7 +12669,7 @@ static int reload_config(enum channelreloadreason reason) global_rtpkeepalive = 0; } } else if (!strcasecmp(v->name, "videosupport")) { - global_videosupport = ast_true(v->value); + ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT); } else if (!strcasecmp(v->name, "compactheaders")) { compactheaders = ast_true(v->value); } else if (!strcasecmp(v->name, "notifymimetype")) { @@ -12757,6 +12821,10 @@ static int reload_config(enum channelreloadreason reason) } } else if (!strcasecmp(v->name, "callevents")) { global_callevents = ast_true(v->value); + } else if (!strcasecmp(v->name, "maxcallbitrate")) { + default_maxcallbitrate = atoi(v->value); + if (default_maxcallbitrate < 0) + default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE; } } |