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Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c11
1 files changed, 6 insertions, 5 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 051bb2bc4..8a7ca5454 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -11095,7 +11095,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
if (framing && p->autoframing) {
ast_debug(1, "Setting framing to %ld\n", framing);
- ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), framing);
+ ast_format_cap_set_framing(p->caps, framing);
}
found = TRUE;
} else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
@@ -13384,6 +13384,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
ast_str_append(&a_audio, 0, "a=maxptime:%d\r\n", max_audio_packet_size);
}
+ if (!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ ast_debug(1, "Setting framing on incoming call: %u\n", min_audio_packet_size);
+ ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), min_audio_packet_size);
+ }
+
if (!doing_directmedia) {
if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
add_ice_to_sdp(p->rtp, &a_audio);
@@ -13676,10 +13681,6 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
add_cc_call_info_to_response(p, &resp);
}
if (p->rtp) {
- if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- ast_debug(1, "Setting framing from config on incoming call\n");
- ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(p->rtp), ast_format_cap_get_framing(p->caps));
- }
ast_rtp_instance_activate(p->rtp);
try_suggested_sip_codec(p);
if (p->t38.state == T38_ENABLED) {