HI Serhat, Is it possible to have a packet capture for the cases you mention.
Regards, On Tue, Nov 1, 2016 at 12:15 PM, Serhat Guler <srtgu...@gmail.com> wrote: > Hi, > > I have a setup as follows: > > IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for > webrtc calls. > > Calls(both audio and video) between to sipml5 clients using firefox web > browser is possible. The session is setup for the calls from sipml5 to > Mercuro, but then there isn't audio flow as the codecs are not compatible. > > Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and > OPUS codecs as firefox but this time the session isn't being setup. Boghe > replies with "Reason: SIP; cause=488; text="Bad content" > " I have seen a similar issue has been mentioned here: > https://github.com/c00lz3r0/boghe/issues/157 but the initial invite > request from sipml5 does have the SDP with media attributes. > > > Any advice or are there any other IMS softphones that I can use to test > for this scenario. Thanks a lot. > > P.S. The previous email went out directly unintentionally. > Serhat > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Alberto Llamas Phone: +1-786-805-6003 Telecommunications Engineer Digium Certified Asterisk Professional (dCap)
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