From 0874e3c825ece4331103441398b4d2db09b72a32 Mon Sep 17 00:00:00 2001 From: Automerge script Date: Sun, 11 Nov 2012 17:20:20 +0000 Subject: [PATCH] Merged revisions 376131 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376131 | file | 2012-11-11 11:15:47 -0600 (Sun, 11 Nov 2012) | 16 lines Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default. With ICE support enabled in chan_sip and a large number of interfaces on the system it was possible for the produced SDP to be truncated due to some fixed size buffers. These buffers have now been changed so they will dynamically grow as needed. ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is no longer enabled by default there. (closes issue ASTERISK-20643) Reported by: coopvr ........ Merged revisions 376130 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 15 ++++++--------- configs/sip.conf.sample | 5 +++-- res/res_rtp_asterisk.c | 2 +- 3 files changed, 10 insertions(+), 12 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 9059669f06..c939df955b 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -12927,9 +12927,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int struct ast_str *m_video = ast_str_alloca(256); /* Media declaration line for video */ struct ast_str *m_text = ast_str_alloca(256); /* Media declaration line for text */ struct ast_str *m_modem = ast_str_alloca(256); /* Media declaration line for modem */ - struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */ - struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */ - struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */ + struct ast_str *a_audio = ast_str_create(256); /* Attributes for audio */ + struct ast_str *a_video = ast_str_create(256); /* Attributes for video */ + struct ast_str *a_text = ast_str_create(256); /* Attributes for text */ struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */ const char *a_crypto = NULL; const char *v_a_crypto = NULL; @@ -13193,11 +13193,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int add_dtls_to_sdp(p->rtp, &a_audio); } - - if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 || - m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 || - a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2) - ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); } if (add_t38) { @@ -13376,6 +13371,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmpcap)); add_sdp_cleanup: + ast_free(a_text); + ast_free(a_video); + ast_free(a_audio); alreadysent = ast_format_cap_destroy(alreadysent); tmpcap = ast_format_cap_destroy(tmpcap); @@ -31227,7 +31225,6 @@ static int reload_config(enum channelreloadreason reason) ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */ ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */ ast_set_flag(&global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT); /*!< Default to nat=auto_force_rport */ - ast_set_flag(&global_flags[2], SIP_PAGE3_ICE_SUPPORT); /*!< Default to enabling ICE support */ ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine)); ast_copy_string(default_parkinglot, DEFAULT_PARKINGLOT, sizeof(default_parkinglot)); diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 3d6db9798e..bbd2af8df4 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -919,10 +919,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; subscribe_network_change_event = yes ; on by default ; -; ICE/STUN/TURN usage can be disabled globally or on a per-peer basis using the icesupport +; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport ; configuration option. When set to yes ICE support is enabled. When set to no it is disabled. +; It is disabled by default. ; -; icesupport = no +; icesupport = yes ;----------------------------------- MEDIA HANDLING -------------------------------- ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index 45df0f9238..7179d592a4 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -111,7 +111,7 @@ enum strict_rtp_state { }; #define DEFAULT_STRICT_RTP STRICT_RTP_CLOSED -#define DEFAULT_ICESUPPORT 0 +#define DEFAULT_ICESUPPORT 1 extern struct ast_srtp_res *res_srtp; extern struct ast_srtp_policy_res *res_srtp_policy;