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-rw-r--r--res/res_pjsip_sdp_rtp.c31
1 files changed, 9 insertions, 22 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 2ecd11a84..7756179b3 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -252,8 +252,8 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
/* get the joint capabilities between peer and endpoint */
ast_format_cap_get_compatible(caps, peer, joint);
if (!ast_format_cap_count(joint)) {
- struct ast_str *usbuf = ast_str_alloca(64);
- struct ast_str *thembuf = ast_str_alloca(64);
+ struct ast_str *usbuf = ast_str_alloca(256);
+ struct ast_str *thembuf = ast_str_alloca(256);
ast_rtp_codecs_payloads_destroy(&codecs);
ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
@@ -269,33 +269,22 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
if (session->channel) {
- struct ast_format *fmt;
-
ast_channel_lock(session->channel);
ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel), AST_MEDIA_TYPE_UNKNOWN);
+ ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
+ AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, media_type);
-
- /*
- * XXX Historically we picked the "best" joint format to use
- * and stuck with it. It would be nice to just append the
- * determined joint media capabilities to give translation
- * more formats to choose from when necessary. Unfortunately,
- * there are some areas of the system where this doesn't work
- * very well. (The softmix bridge in particular is reluctant
- * to pick higher fidelity formats and has a problem with
- * asymmetric sample rates.)
- */
- fmt = ast_format_cap_get_format(joint, 0);
- ast_format_cap_append(caps, fmt, 0);
+ ast_format_cap_append_from_cap(caps, joint, media_type);
/*
* Apply the new formats to the channel, potentially changing
* raw read/write formats and translation path while doing so.
*/
ast_channel_nativeformats_set(session->channel, caps);
- ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
- ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
+ if (media_type == AST_MEDIA_TYPE_AUDIO) {
+ ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
+ ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
+ }
if ((session->endpoint->dtmf == AST_SIP_DTMF_AUTO)
&& (ast_rtp_instance_dtmf_mode_get(session_media->rtp) == AST_RTP_DTMF_MODE_RFC2833)
&& (session->dsp)) {
@@ -309,8 +298,6 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
}
}
ast_channel_unlock(session->channel);
-
- ao2_ref(fmt, -1);
}
ast_rtp_codecs_payloads_destroy(&codecs);