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Move check for codec translators to an earlier place in the call, so we can fail gracefully (imported from 1.4) 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2006-11-02 20:24:10 +00:00
parent 0df2c7a774
commit 9dffcc2e75
1 changed files with 20 additions and 17 deletions
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37
channels/chan_sip.c
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@ -1279,7 +1279,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
int debug);
static int add_sdp(struct sip_request *resp, struct sip_pvt *p);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p);
static void do_setnat(struct sip_pvt *p, int natflags);
/*--- Authentication stuff */
@ -2870,12 +2870,21 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
res = update_call_counter(p, INC_CALL_RINGING);
if ( res != -1 ) {
p->callingpres = ast->cid.cid_pres;
p->jointcapability = p->capability;
p->t38.jointcapability = p->t38.capability;
if (option_debug)
ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
/* If there are no audio formats left to offer, punt */
if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
res = -1;
} else {
p->t38.jointcapability = p->t38.capability;
if (option_debug > 1)
ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
}
}
return res;
}
@ -6027,7 +6036,7 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
}
/*! \brief Add Session Description Protocol message */
static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
{
int len = 0;
int alreadysent = 0;
@ -6069,7 +6078,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return -1;
return AST_FAILURE;
}
/* Set RTP Session ID and version */
@ -6093,14 +6102,8 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
dest.sin_port = sin.sin_port;
}
/* Ok, let's start working with codec selection here */
capability = ast_translate_available_formats(p->jointcapability, p->prefcodec);
capability = p->jointcapability;
/* If there are no audio formats left to offer, punt */
if (!(capability & AST_FORMAT_AUDIO_MASK)) {
ast_log(LOG_WARNING, "No audio format found to offer.\n");
return -1;
}
if (option_debug > 1) {
char codecbuf[BUFSIZ];
@ -6282,7 +6285,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
}
return 0;
return AST_SUCCESS;
}
/*! \brief Used for 200 OK and 183 early media */