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chan_sip: Fix autoframing=yes.

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With Asterisk 13, the structures ast_format and ast_codec changed. Because of
that, the paketization timing (framing) of the RTP channel moved away from the
formats/codecs. In the course of that change, the ptime of the callee was not
honored anymore, when the optional autoframing was enabled.

ASTERISK-25484 #close

Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4
This commit is contained in:
Alexander Traud
2015-10-21 16:51:11 +02:00
parent 7c560a9a31
commit 869ef2a8ee
1 changed files with 6 additions and 5 deletions
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11
channels/chan_sip.c
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@@ -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) {