today i replaced mediaproxy-ng with rtpengine. then i made a test call
from baresip to jssip web ua. rtpengine module gave this kind of offer
to rtpengine:
Apr 7 10:56:34 siika /usr/sbin/sip-proxy[24013]: INFO: ===== rtpengine_offer(via-branch=1 ICE=force trust-address)
and this kind of answer:
Apr 7 11:07:18 siika /usr/sbin/sip-proxy[24013]: INFO: ===== rtpengine_answer(replace-session-connection replace-origin via-branch=2 trust-address)
there was problem with media (baresip console output included below).
the same test call works fine with mediaproxy-ng, i.e., everything else
in the test is same except that mediaproxy-ng is replaced with rtpproxy.
-- juha
------------------------------------------------------------------
call: connecting to 'sip:test@test.tutpro.com'..
ice: audio: Default local candidates: ? / ?
call: SIP Progress: 100 Trying ()
call: SIP Progress: 100 Trying ()
call: SIP Progress: 180 Ringing ()
{ audio.1} Set default remote candidate: host:192.98.102.30:50002
{ audio.2} Set default remote candidate: host:192.98.102.30:50003
audio: Set audio decoder: opus 48000Hz 2ch
audio: Set audio encoder: opus 48000Hz 2ch
audio tx pipeline: alsa ---> opus
audio rx pipeline: alsa <--- opus
jh(a)test.tutpro.com: Call established: sip:test@test.tutpro.com
opus: decode error: buffer too small
audio: opus codec decode 273 bytes: Protocol error
opus: decode error: buffer too small
audio: opus codec decode 289 bytes: Protocol error
opus: decode error: corrupted stream
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: buffer too small
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: buffer too small
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: buffer too small
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: corrupted stream
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: corrupted stream
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: buffer too small
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: corrupted stream
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: buffer too small
audio: opus codec decode 323 bytes: Protocol error
opus: decode error: corrupted stream
...
Hi,
I've been having problems with mi-xmlrpc module in Kamailio 4.0.6 + Debian
Wheezy (7.4)
The error seems to be related with libxmlrpc-c3 version.
root@debian:/usr/local/src/kamailio/modules/mi_xmlrpc# make
Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.33),
compile at your own risk!
Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.33),
compile at your own risk!
CC (gcc) [M mi_xmlrpc.so] abyss_conf.o
CC (gcc) [M mi_xmlrpc.so] abyss_conn.o
CC (gcc) [M mi_xmlrpc.so] abyss_data.o
CC (gcc) [M mi_xmlrpc.so] abyss_date.o
CC (gcc) [M mi_xmlrpc.so] abyss_file.o
CC (gcc) [M mi_xmlrpc.so] abyss_http.o
abyss_http.c: In function 'RequestInit':
abyss_http.c:105:5: warning: implicit declaration of function 'StringAlloc'
[-Wimplicit-function-declaration]
abyss_http.c: In function 'RequestFree':
abyss_http.c:120:5: warning: implicit declaration of function 'StringFree'
[-Wimplicit-function-declaration]
CC (gcc) [M mi_xmlrpc.so] abyss_response.o
abyss_response.c:590:6: error: conflicting types for 'Base64Encode'
In file included from abyss_response.c:20:0:
/usr/include/xmlrpc-c/abyss.h:513:1: note: previous declaration of
'Base64Encode' was here
make: *** [abyss_response.o] Error 1
Some months ago was a commit related with that (I think so):
http://lists.sip-router.org/pipermail/sr-dev/2013-October/021513.html
It was a patch witten by Muhammad Shahzad and was applied in 4.0.4 version,
but seems to be incomplete.
I've solved the error, patching in the same way "abyss_response.c" (there
are still some warnings)
--- abyss_response.c 2014-03-26 16:57:07.468091972 +0100
+++ abyss_response.c 2014-03-06 16:40:16.000000000 +0100
@@ -582,6 +582,8 @@
}
+#ifdef XMLRPC_OLD_VERSION
/*********************************************************************
** Base64
*********************************************************************/
@@ -622,6 +624,7 @@
/* ...and zero-terminate it. */
*p = '\0';
}
+#endif
/******************************************************************************
**
In 4.1.2 version compiles OK (I've backported the fix from here)
Cheers,
P.D.: Kamailio 4.0 deb packages have same problem I guess, cause there is
no mi_xmlrpc.so
--
Andoni Izurza
You got it!! ;P
before switching to mediaproxy-ng, i was using mediaproxy and kamailio
mediaproxy module. when it added ice candidate, candidate type was
'relay'. now i noticed that rtpengine adds candidate of type 'host' and
there seems to be no offer argument to change that.
i'm thinking of scenario where parties of the call may be able to
find out a way to communicate directly in which case rtpengine should
not be forced on them. that could be achieved by adding 'relay' type
candidate to sdp instead of 'host' type candidate.
what is rtpengine motivation for adding 'host' type candidate instead of
'relay' type candidate?
-- juha
THIS IS AN AUTOMATED MESSAGE, DO NOT REPLY.
The following task is now closed:
FS#415 - module cdp fails to start - undefined symbol: sem_post + patch
User who did this - Jason Penton (jaybeepee)
Reason for closing: Not a bug
Additional comments about closing: already patched in 4.2 and master
More information can be found at the following URL:
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The following task has a new comment added:
FS#415 - module cdp fails to start - undefined symbol: sem_post + patch
User who did this - Leon de Rooij (leonderooij)
----------
aah, thanks, I'm on 4.1 branch - I see it's fixed in master :-)
regards,
Leon
----------
More information can be found at the following URL:
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The following task has a new comment added:
FS#415 - module cdp fails to start - undefined symbol: sem_post + patch
User who did this - Jason Penton (jaybeepee)
----------
Hi Leon, Which version are you using? On master it is already like that?
----------
More information can be found at the following URL:
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