Hi, Jeff!
Unfortunately I don't think this will work.
The idea is that during a message processing, the same SDP buffer is
seen by all functions. Changes do not immediately alter the buffer, but
they are registered as lumps (changes that are later applied), when the
message processing is done. So even in the branch route you will see the
initial SDP, because it is part of the same processing context (branch
route is triggered by the t_relay() function).
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 04/18/2017 09:44 PM, Jeff Pyle wrote:
Dragomir,
If Zoiper speaks only G.729, and SIP.js speaks only G.711, rtpengine
isn't going to help. It doesn't transcode. From its github page
<https://github.com/sipwise/rtpengine>:
/Rtpengine/ does not (yet) support:
o Repacketization or transcoding
Is iLBC an option for you in SIP.js and Zoiper? It's license free and
sounds a little bitter. If not, Asterisk or FreeSWITCH could perform
this task with the appropriate G.729 licenses.
Răzvan,
Is there any effect of using either the codec manipulation or
rtpengine in a branch route? I ask this admittedly not understanding
the buffers in use.
- Jeff
On Tue, Apr 18, 2017 at 12:39 PM, Dragomir Haralambiev
<goup2...@gmail.com <mailto:goup2...@gmail.com>> wrote:
Hi Razvan,
How to make follow connection using rtpengine?
Zoiper(g729) <-----> Opensips(rtpengine) <--------> browser
(SIP.JS with g711)
2017-04-18 19:10 GMT+03:00 Răzvan Crainea <raz...@opensips.org
<mailto:raz...@opensips.org>>:
Hi, Jeff!
Unfortunately you can't use both rtpengine and codec_delete_*,
that's because each change different buffers. The
codec_delete_* function runs on the initial SDP received, then
rtpengine completely overwrites the SDP with whatever
rtpengine replied.
The only way you can do something like this (although it may
be very ugly) is to store the rtpengine reply in a pvar using
the 3rd[1] parameter of the rtpengine_* functions and perform
some text replaces[2] on it, then replace the body "manually".
[1]
http://www.opensips.org/html/docs/modules/2.3.x/rtpengine.html#rtpengine.f.rtpengine_offer
<http://www.opensips.org/html/docs/modules/2.3.x/rtpengine.html#rtpengine.f.rtpengine_offer>
[2]
http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728
<http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728>
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com <http://www.opensips-solutions.com>
On 04/18/2017 06:49 PM, Jeff Pyle wrote:
Hello,
This is on OpenSIPS 2.3, downloaded from git and compiled today.
An INVITE arrives over TLS with the following SDP:
v=0
o=- 1492528621 1492528621 IN IP4 172.22.202.191
s=Polycom IP Phone
c=IN IP4 172.22.202.191
t=0 0
m=audio 16852 RTP/SAVP 115 9 0 8 110 18 127
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 iLBC/8000
a=fmtp:110 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
a=rtcp:16853
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:[stripped]
a=setup:actpass
a=fingerprint:sha-1 [stripped]
m=audio 16888 RTP/AVP 115 9 0 8 110 18 127
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 iLBC/8000
a=fmtp:110 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
a=rtcp:16889
I run
codec_delete_expect_re(PCMU|PCMA|telephone-event)
but it doesn't have any effect. The INVITE leaving after
t_relay() over UDP to localhost on a different port is the
same as when it came in (with the exception of the c= line
because of rtpengine).
At log_level=6 the only log entry I see is
DBG:sipmsgops:create_codec_lumps: creating 0 streams
I'm not sure where to go from here.
- Jeff
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