Displaying 5 results from an estimated 5 matches for "payloadtyp".
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payloadtype
2003 May 30
1
siemens optipoint 400 SIP
hi!
anyone try siemens optipoint 400 economy SIP phone with * ?
--
http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf
Thomas
2003 Jul 08
1
RTP.C codec error 19
hi ..
when placing a SIP call to a sip host in the states
every few seconds I get an RTP codec 19 error.
I know this is related to comfort noise, and the
call goes through OK ... how can I suppress
the error message ?
Also, many times I get "Invalid CSeq Number"
back from 216.52.153.207 (which is the host
i'm calling) and the call drops.. is there a solution
for this ?
cheers
Dave
2003 Feb 23
0
Question about some Cisco-specific code in "rtp.c"
The file "rtp.c" currently contains the following hack for special-case
handling of some Cisco-specific protocol:
} else if (payloadtype == 121) {
/* CISCO proprietary DTMF bridge */
f = process_type121(rtp, rtp->rawdata +
AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
As I noted in my earlier message, I'm planning to update the code to
improve the handling of dynamic RTP pa...
2005 Sep 16
0
lastest spandsp-0.03pre1 don't compile
..._test_addr' follows
non-static declaration
udptl.c:541: warning: 'udptl_debug_test_addr' declared inline after being
called
udptl.c:541: warning: previous implicit declaration of
'udptl_debug_test_addr' was here
udptl.c: In function `ast_udptl_read':
udptl.c:643: error: `payloadtype' undeclared (first use in this function)
udptl.c:643: error: `timestamp' undeclared (first use in this function)
udptl.c:643: error: `hdrlen' undeclared (first use in this function)
udptl.c:649: error: `AST_FORMAT_T38' undeclared (first use in this function)
udptl.c: In functio...
2017 Apr 26
3
pjsip direct_media=yes and "unknown" endpoints
I'm trying to implement direct_media between multiple peers and an
uplink provider, all of whom have direct_media=yes configures.
For originating calls to the uplink provider direct_media=yes works like
expected. SIP flows through asterisk, rtp doesn't
SIP: enduser <-> SBC <-> asterisk 13 <-> uplink
RTP: enduser <-> SBC <-----------------> uplink
SBC