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