From b54dda4cefe89665d98074cf3767858fa6ecdb6e Mon Sep 17 00:00:00 2001 From: Sylvain Munaut Date: Sun, 20 Dec 2009 22:06:40 +0100 Subject: [PATCH] ipaccess: Send RTP Payload IE for CRCX & MDCX For GSM V1 FR, the payload type is fixed to 3 in the RFC. But for the other codecs, the payload type is dynamically assigned between 96 and 127. Here, we use a static mapping internal to OpenBSC. This patch is needed to make a rather old 139 unit (with sw version 120a002_v149b42d0) work with something else than FR codec. I also tested this patch on a newer 139 (with sw version 120a352_v267b22d0) to make sure it didn't add a regression. More testing with newer EDGE units should be done by whoever has some of theses. Signed-off-by: Sylvain Munaut --- openbsc/include/openbsc/gsm_data.h | 1 + openbsc/include/openbsc/rtp_proxy.h | 6 ++++ openbsc/src/abis_rsl.c | 54 ++++++++++++++++++++++++++--- openbsc/src/rtp_proxy.c | 3 -- 4 files changed, 56 insertions(+), 8 deletions(-) diff --git a/openbsc/include/openbsc/gsm_data.h b/openbsc/include/openbsc/gsm_data.h index ef0b50768..68d3deebf 100644 --- a/openbsc/include/openbsc/gsm_data.h +++ b/openbsc/include/openbsc/gsm_data.h @@ -250,6 +250,7 @@ struct gsm_lchan { u_int16_t bound_port; u_int16_t connect_port; u_int16_t conn_id; + u_int8_t rtp_payload; u_int8_t rtp_payload2; u_int8_t speech_mode; struct rtp_socket *rtp_socket; diff --git a/openbsc/include/openbsc/rtp_proxy.h b/openbsc/include/openbsc/rtp_proxy.h index f82711a8e..65b1a5fac 100644 --- a/openbsc/include/openbsc/rtp_proxy.h +++ b/openbsc/include/openbsc/rtp_proxy.h @@ -28,6 +28,12 @@ #include #include +#define RTP_PT_GSM_FULL 3 +#define RTP_PT_GSM_HALF 96 +#define RTP_PT_GSM_EFR 97 +#define RTP_PT_AMR_FULL 98 +#define RTP_PT_AMR_HALF 99 + enum rtp_rx_action { RTP_NONE, RTP_PROXY, diff --git a/openbsc/src/abis_rsl.c b/openbsc/src/abis_rsl.c index ba1be99c8..343347aa5 100644 --- a/openbsc/src/abis_rsl.c +++ b/openbsc/src/abis_rsl.c @@ -1368,6 +1368,44 @@ static u_int8_t ipa_smod_s_for_lchan(struct gsm_lchan *lchan) return 0; } +static u_int8_t ipa_rtp_pt_for_lchan(struct gsm_lchan *lchan) +{ + switch (lchan->tch_mode) { + case GSM48_CMODE_SPEECH_V1: + switch (lchan->type) { + case GSM_LCHAN_TCH_F: + return RTP_PT_GSM_FULL; + case GSM_LCHAN_TCH_H: + return RTP_PT_GSM_HALF; + default: + break; + } + case GSM48_CMODE_SPEECH_EFR: + switch (lchan->type) { + case GSM_LCHAN_TCH_F: + return RTP_PT_GSM_EFR; + /* there's no half-rate EFR */ + default: + break; + } + case GSM48_CMODE_SPEECH_AMR: + switch (lchan->type) { + case GSM_LCHAN_TCH_F: + return RTP_PT_AMR_FULL; + case GSM_LCHAN_TCH_H: + return RTP_PT_AMR_HALF; + default: + break; + } + default: + break; + } + LOGP(DRSL, LOGL_ERROR, "Cannot determine ip.access rtp payload type for " + "tch_mode == 0x%02x\n & lchan_type == %d", + lchan->tch_mode, lchan->type); + return 0; +} + /* ip.access specific RSL extensions */ static void ipac_parse_rtp(struct gsm_lchan *lchan, struct tlv_parsed *tv) { @@ -1434,10 +1472,13 @@ int rsl_ipacc_crcx(struct gsm_lchan *lchan) /* 0x1- == receive-only, 0x-1 == EFR codec */ lchan->abis_ip.speech_mode = 0x10 | ipa_smod_s_for_lchan(lchan); + lchan->abis_ip.rtp_payload = ipa_rtp_pt_for_lchan(lchan); msgb_tv_put(msg, RSL_IE_IPAC_SPEECH_MODE, lchan->abis_ip.speech_mode); + msgb_tv_put(msg, RSL_IE_IPAC_RTP_PAYLOAD, lchan->abis_ip.rtp_payload); - DEBUGP(DRSL, "%s IPAC_BIND speech_mode=0x%02x\n", - gsm_lchan_name(lchan), lchan->abis_ip.speech_mode); + DEBUGP(DRSL, "%s IPAC_BIND speech_mode=0x%02x RTP_PAYLOAD=%d\n", + gsm_lchan_name(lchan), lchan->abis_ip.speech_mode, + lchan->abis_ip.rtp_payload); msg->trx = lchan->ts->trx; @@ -1464,11 +1505,13 @@ int rsl_ipacc_mdcx(struct gsm_lchan *lchan, u_int32_t ip, u_int16_t port, /* 0x0- == both directions, 0x-1 == EFR codec */ lchan->abis_ip.speech_mode = 0x00 | ipa_smod_s_for_lchan(lchan); + lchan->abis_ip.rtp_payload = ipa_rtp_pt_for_lchan(lchan); ia.s_addr = htonl(ip); - DEBUGP(DRSL, "%s IPAC_MDCX IP=%s PORT=%d RTP_PAYLOAD2=%d CONN_ID=%d " - "speech_mode=0x%02x\n", gsm_lchan_name(lchan), inet_ntoa(ia), port, - rtp_payload2, lchan->abis_ip.conn_id, lchan->abis_ip.speech_mode); + DEBUGP(DRSL, "%s IPAC_MDCX IP=%s PORT=%d RTP_PAYLOAD=%d RTP_PAYLOAD2=%d " + "CONN_ID=%d speech_mode=0x%02x\n", gsm_lchan_name(lchan), + inet_ntoa(ia), port, lchan->abis_ip.rtp_payload, rtp_payload2, + lchan->abis_ip.conn_id, lchan->abis_ip.speech_mode); msgb_tv16_put(msg, RSL_IE_IPAC_CONN_ID, lchan->abis_ip.conn_id); msgb_v_put(msg, RSL_IE_IPAC_REMOTE_IP); @@ -1476,6 +1519,7 @@ int rsl_ipacc_mdcx(struct gsm_lchan *lchan, u_int32_t ip, u_int16_t port, *att_ip = ia.s_addr; msgb_tv16_put(msg, RSL_IE_IPAC_REMOTE_PORT, port); msgb_tv_put(msg, RSL_IE_IPAC_SPEECH_MODE, lchan->abis_ip.speech_mode); + msgb_tv_put(msg, RSL_IE_IPAC_RTP_PAYLOAD, lchan->abis_ip.rtp_payload); if (rtp_payload2) msgb_tv_put(msg, RSL_IE_IPAC_RTP_PAYLOAD2, rtp_payload2); diff --git a/openbsc/src/rtp_proxy.c b/openbsc/src/rtp_proxy.c index 375204e97..924173dd2 100644 --- a/openbsc/src/rtp_proxy.c +++ b/openbsc/src/rtp_proxy.c @@ -91,9 +91,6 @@ struct rtp_x_hdr { #define RTP_VERSION 2 -#define RTP_PT_GSM_FULL 3 -#define RTP_PT_GSM_EFR 97 - /* decode an rtp frame and create a new buffer with payload */ static int rtp_decode(struct msgb *msg, u_int32_t callref, struct msgb **data) {