diff options
Diffstat (limited to 'res')
| -rw-r--r-- | res/res_pjsip.c | 9 | ||||
| -rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 1 | ||||
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 17 |
3 files changed, 21 insertions, 6 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index ff18a1d6d..27e3f8179 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -471,6 +471,15 @@ set to <literal>sdes</literal> or <literal>dtls</literal>. </para></description> </configOption> + <configOption name="g726_non_standard" default="no"> + <synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis> + <description><para> + When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2 + packing order instead of what is recommended by RFC3551. Since this essentially + replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be + specified in the endpoint's allowed codec list. + </para></description> + </configOption> <configOption name="inband_progress" default="no"> <synopsis>Determines whether chan_pjsip will indicate ringing using inband progress.</synopsis> diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index 34bb69211..9a383a25a 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1922,6 +1922,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "dtls_fingerprint", "", dtls_handler, dtlsfingerprint_to_str, NULL, 0, 0); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "srtp_tag_32", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.srtp_tag_32)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "media_encryption_optimistic", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.encryption_optimistic)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "g726_non_standard", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.g726_non_standard)); ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "redirect_method", "user", redirect_handler, NULL, NULL, 0, 0); ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "set_var", "", set_var_handler, set_var_to_str, set_var_to_vl, 0, 0); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "message_context", "", OPT_STRINGFIELD_T, 1, STRFLDSET(struct ast_sip_endpoint, message_context)); diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 1bd3b0cb3..fb70dd331 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -155,6 +155,8 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp char name[256]; char media[20]; char fmt_param[256]; + enum ast_rtp_options options = session->endpoint->media.g726_non_standard ? + AST_RTP_OPT_G726_NONSTANDARD : 0; ast_rtp_codecs_payloads_initialize(codecs); @@ -176,9 +178,10 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp if (strcmp(name,"telephone-event") == 0) { tel_event++; } + ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media)); ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]), - media, name, 0, rtpmap->clock_rate); + media, name, options, rtpmap->clock_rate); /* Look for an optional associated fmtp attribute */ if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) { continue; @@ -304,18 +307,20 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi return 0; } -static pjmedia_sdp_attr* generate_rtpmap_attr(pjmedia_sdp_media *media, pj_pool_t *pool, int rtp_code, - int asterisk_format, struct ast_format *format, int code) +static pjmedia_sdp_attr* generate_rtpmap_attr(struct ast_sip_session *session, pjmedia_sdp_media *media, pj_pool_t *pool, + int rtp_code, int asterisk_format, struct ast_format *format, int code) { pjmedia_sdp_rtpmap rtpmap; pjmedia_sdp_attr *attr = NULL; char tmp[64]; + enum ast_rtp_options options = session->endpoint->media.g726_non_standard ? + AST_RTP_OPT_G726_NONSTANDARD : 0; snprintf(tmp, sizeof(tmp), "%d", rtp_code); pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp); rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1]; rtpmap.clock_rate = ast_rtp_lookup_sample_rate2(asterisk_format, format, code); - pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, 0)); + pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options)); rtpmap.param.slen = 0; rtpmap.param.ptr = NULL; @@ -1050,7 +1055,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as continue; } - if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 1, format, 0))) { + if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) { ao2_ref(format, -1); continue; } @@ -1075,7 +1080,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as continue; } - if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 0, NULL, index))) { + if (!(attr = generate_rtpmap_attr(session, media, pool, rtp_code, 0, NULL, index))) { continue; } |