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authorHolger Hans Peter Freyther <zecke@selfish.org>2011-08-22 18:22:44 +0200
committerHolger Hans Peter Freyther <zecke@selfish.org>2011-08-22 18:22:44 +0200
commit128892f55f23ac24e6eb786ca3cf0841fbc45fd2 (patch)
tree866fc7fee30cd3d914f881a6900cbd044ea613f1 /openbsc/src/osmo-bsc/osmo_bsc_audio.c
parent3488c3ddc46d74b7b2bbc879692dbf6e7ad2ecc3 (diff)
audio: Remove the hardcoding of the RTP Payload
The MGCP config must be correct and use 99 for RTP AMR.
Diffstat (limited to 'openbsc/src/osmo-bsc/osmo_bsc_audio.c')
-rw-r--r--openbsc/src/osmo-bsc/osmo_bsc_audio.c1
1 files changed, 0 insertions, 1 deletions
diff --git a/openbsc/src/osmo-bsc/osmo_bsc_audio.c b/openbsc/src/osmo-bsc/osmo_bsc_audio.c
index 64260aa88..ed0ece761 100644
--- a/openbsc/src/osmo-bsc/osmo_bsc_audio.c
+++ b/openbsc/src/osmo-bsc/osmo_bsc_audio.c
@@ -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;
}