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osmo-bsc
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audio: Remove the hardcoding of the RTP Payload 

The MGCP config must be correct and use 99 for RTP AMR.
This commit is contained in:
Holger Hans Peter Freyther 2011-08-22 18:22:44 +02:00
parent 3488c3ddc4
commit 128892f55f
3 changed files with 0 additions and 6 deletions
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1
openbsc/include/openbsc/gsm_data.h
View File

@ -284,7 +284,6 @@ struct gsm_network {
/* MSC data in case we are a true BSC */
struct osmo_msc_data *msc_data;
int hardcoded_rtp_payload;
/* subscriber related features */
int keep_subscr;

4
openbsc/src/libbsc/abis_rsl.c
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@ -1580,10 +1580,6 @@ static uint8_t ipa_rtp_pt_for_lchan(struct gsm_lchan *lchan)
{
struct gsm_network *net = lchan->ts->trx->bts->network;
/* allow to hardcode the rtp payload */
if (net->hardcoded_rtp_payload != 0)
return net->hardcoded_rtp_payload;
switch (lchan->tch_mode) {
case GSM48_CMODE_SPEECH_V1:
switch (lchan->type) {

1
openbsc/src/osmo-bsc/osmo_bsc_audio.c
View File

@ -64,7 +64,6 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
int osmo_bsc_audio_init(struct gsm_network *net)
{
net->hardcoded_rtp_payload = 98;
osmo_signal_register_handler(SS_ABISIP, handle_abisip_signal, net);
return 0;
}