Sorry, I should've included that in the original mesg... The logs I posted were from trunk-r10055. I updated this morning to trunk-r10081, still the same issue.
~Gabe Brian West wrote: > Which version of FreeSWITCH are you using? > > /b > > On Oct 20, 2008, at 2:25 AM, Gabriel Kuri wrote: > >> I'm having an issue with the linksys spa devices when enabling inbound >> proxy media mode (inbound-proxy-media=true) and late negotiation >> (inbound-late-negotiation=true) in the sofia profile. The spa >> immediately sends a BYE when the call is answered by the called party. >> For whatever reason, it works fine between two linksys devices >> directly >> connected to FS, but when the call goes out to the PSTN via the SIP >> provider, the spa isn't happy and sends a BYE. >> >> After comparing the raw SIP packets on the wire (tcpdump) and between >> enabling/disabling proxy-media mode and late negotiation, the only >> difference I notice is the port in the m= line of the SDP header. >> >> According to the freeswitch log, the rtp port would be rewritten to >> 28044 in the sdp header of the SIP packet sent to the spa device. >> But on >> the wire, the port is rewritten to 0, which I'm guessing is why the >> spa >> isn't happy and sending a BYE. >> >> Here's the excerpt from the freeswitch log showing FS rewriting the >> port >> to 28044 for the packet going to the spa device. >> >> >> [DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp() >> sofia/internal/<phone_number_removed>@mydomain.net Patched SDP >> --- >> v=0 >> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX >> s=sip call >> c=IN IP4 XX.XX.XX.XX >> t=0 0 >> m=audio 24174 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> +++ >> v=0 >> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX >> s=sip call >> c=IN IP4 YY.YY.YY.YY >> t=0 0 >> m=audio 28044 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> >> However, the packet on the wire reveals FS rewriting the port to >> 0 ... >> >> v=0. >> o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 >> YY.YY.YY.YY. >> s=sip call. >> c=IN IP4 YY.YY.YY.YY. >> t=0 0. >> m=audio 0 RTP/AVP 96 101. >> a=rtpmap:96 G729/8000. >> a=fmtp:96 annexb=no. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-15. >> >> Is this a bug or is there some other problem? >> >> Thanks for the help, >> Gabe >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> [email protected] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
