diff options
| -rw-r--r-- | CHANGES | 4 | ||||
| -rw-r--r-- | channels/chan_sip.c | 58 |
2 files changed, 44 insertions, 18 deletions
@@ -553,6 +553,10 @@ chan_sip set of proxies by using a pre-loaded route-set defined by the Path headers in the REGISTER request. See Realtime updates for more configuration information. + * The SIP_CODEC family of variables may now specify more than one codec. Each + codec must be separated by a comma. The first codec specified is the + preferred codec for the offer. This allows a dialplan writer to specify both + audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264) Functions ------------------ diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 2e7615ae7..519f1dc88 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -7350,35 +7350,57 @@ static int sip_hangup(struct ast_channel *ast) return 0; } -/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */ +/*! \brief Try setting the codecs suggested by the SIP_CODEC channel variable */ static void try_suggested_sip_codec(struct sip_pvt *p) { struct ast_format fmt; - const char *codec; + const char *codec_list; + char *codec_list_copy; + struct ast_format_cap *original_jointcaps; + char *codec; + int first_codec = 1; - ast_format_clear(&fmt); + char *strtok_ptr; if (p->outgoing_call) { - codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND"); - } else if (!(codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) { - codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC"); + codec_list = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND"); + } else if (!(codec_list = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) { + codec_list = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC"); } - if (!codec) + if (ast_strlen_zero(codec_list)) { return; + } - ast_getformatbyname(codec, &fmt); - if (fmt.id) { - ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec); - if (ast_format_cap_iscompatible(p->jointcaps, &fmt)) { - ast_format_cap_set(p->jointcaps, &fmt); - ast_format_cap_set(p->caps, &fmt); - } else - ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n"); - } else - ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec); + codec_list_copy = ast_strdupa(codec_list); + original_jointcaps = ast_format_cap_dup(p->jointcaps); + + for (codec = strtok_r(codec_list_copy, ",", &strtok_ptr); codec; codec = strtok_r(NULL, ",", &strtok_ptr)) { + codec = ast_strip(codec); + + if (!ast_getformatbyname(codec, &fmt)) { + ast_log(AST_LOG_NOTICE, "Ignoring ${SIP_CODEC*} variable because of unrecognized/not configured codec %s (check allow/disallow in sip.conf)\n", codec); + continue; + } + if (ast_format_cap_iscompatible(original_jointcaps, &fmt)) { + if (first_codec) { + ast_verb(4, "Set codec to '%s' for this call because of ${SIP_CODEC*} variable\n", codec); + ast_format_cap_set(p->jointcaps, &fmt); + ast_format_cap_set(p->caps, &fmt); + first_codec = 0; + } else { + ast_verb(4, "Add codec to '%s' for this call because of ${SIP_CODEC*} variable\n", codec); + ast_format_cap_add(p->jointcaps, &fmt); + ast_format_cap_add(p->caps, &fmt); + } + } else { + ast_log(AST_LOG_NOTICE, "Ignoring ${SIP_CODEC*} variable because it is not shared by both ends: %s\n", codec); + } + } + ast_format_cap_destroy(original_jointcaps); return; -} + } + /*! \brief sip_answer: Answer SIP call , send 200 OK on Invite * Part of PBX interface */ |