It's an spa issue but we work around it so that shouldn't matter! /b
Sent from my iPhone On Oct 20, 2008, at 9:59 PM, Gabriel Kuri <[EMAIL PROTECTED]> wrote: > I ran into this posting which is similar, although not exactly the > same, > as the failure mode I'm experiencing. > > http://bugs.digium.com/view.php?id=11483 > > following what this other person tried to temporarily fix the issue, I > changed the name of the rtpmap on the linksys spa from G729a to G729 > and > it works - FS no longer transmits an audio port of 0 in the sdp > headers > when inbound-proxy-media and late-negotiation are enabled. > > correct sdp header excerpt on a call ... > > v=0. > o=01Nextone 2341985734634606731 5798373005113647141 IN IP4 > YY.YY.YY.YY. > s=sip call. > c=IN IP4 YY.YY.YY.YY. > t=0 0. > m=audio 25454 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. > > So is this an underlying issue with the linksys spa units or FS? > > Gabe > > > 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
