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- Add some comments to t.38 code 

- Remove improper blocking of ptime: in SDP


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2006-11-12 15:14:35 +00:00
parent e19fc06e2c
commit 1f9dcf758c
1 changed files with 14 additions and 4 deletions
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18
channels/chan_sip.c
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@ -4931,10 +4931,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
if (debug)
ast_verbose("Got unsupported a:crypto in SDP offer \n");
breakout = TRUE;
} else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
if (debug)
ast_verbose("Got unsupported a:ptime in SDP offer \n");
breakout = TRUE;
}
if (breakout) /* We have a match, skip to next header */
continue;
@ -16561,6 +16557,11 @@ static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
return 0;
}
/*! \brief Handle T38 reinvite
\todo Make sure we don't destroy the call if we can't handle the re-invite.
Nothing should be changed until we have processed the SDP and know that we
can handle it.
*/
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite)
{
struct sip_pvt *p;
@ -16572,12 +16573,21 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt
/* Setup everything on the other side like offered/responded from first side */
sip_pvt_lock(p);
/*! \todo check if this is not set earlier when setting up the PVT. If not
maybe it should move there. */
p->t38.jointcapability = p->t38.peercapability = pvt->t38.jointcapability;
ast_udptl_set_far_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
ast_udptl_set_local_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
ast_udptl_set_error_correction_scheme(p->udptl, ast_udptl_get_error_correction_scheme(pvt->udptl));
if (reinvite) { /* If we are handling sending re-invite to the other side of the bridge */
/*! \note The SIP_CAN_REINVITE flag is for RTP media redirects,
not really T38 re-invites which are different. In this
case it's used properly, to see if we can reinvite over
NAT
*/
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) {
ast_udptl_get_peer(pvt->udptl, &p->udptlredirip);
flag =1;