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 <[email protected]>: > 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 > > > > _______________________________________________ > Users mailing > [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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