[OpenSIPS-Users] codec_delete_except_re() has no effect

[Jeff Pyle]

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:


   - 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 <goup2010 at 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 <razvan at 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.ht
>> ml#rtpengine.f.rtpengine_offer
>> [2] http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutionswww.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
>>
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
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