[OpenSIPS-Users] codec_delete_except_re() has no effect
Dragomir Haralambiev
goup2010 at gmail.com
Tue Apr 18 12:39:42 EDT 2017
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.
> html#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
>
>
>
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