On Mon, Nov 24, 2008 at 12:57 PM, Dale Worley <[EMAIL PROTECTED]> wrote: > On Mon, 2008-11-24 at 12:07 -0500, M. Ranganathan wrote: >> Attached is the call flow for a caller initiated Blind Transfer. The >> ITSP is bt.com. >> >> I am wondering about why the ITSP would return 503. The re-INVITE >> looks ok to me and the codecs offered are a subset of the previous >> answer from the ITSP ( but maybe I have been staring too long). A >> second pair of eyes may spot something. > > According to 3261, the 503 response means some sort of overload or > failure at the server, and doesn't depend on the request in question. > So if BT is using 503 correctly (which I would not assume), it's not > *your* problem. I would test it again and then contact BT tech support > to find out what is wrong with the server. > > However, I do see one difference between the initial SDP and the revised > SDP: > > a=rtpmap:0 PCMU/8000 > > vs.: > > a=rtpmap:0 pcmu/8000/1 > > The case of the letters is different (which shouldn't matter if SDP is > case-insensitive) and the second one has "/1" (which I think is the > default, so it shouldn't matter, but it might). > > Dale > > >
Dale, Tried your suggestion and it did not work. It seems to accept media re-negotiation only if you send an sdp answer on the ACK first. So the attached call flow worked out. Note that there has been a change in the way I do SDP answer in ACK. I have to ACK right away with the previous answer and then re-play the answer on the sipx side in a re-INVITE with SDP. Otherwise, the blind transfer times out while the phone is ringing at the transfer target. If you have time, please look at the attached call flow and see if you find any gotcha's with it. BT.COM : By experimentation I have been able to conclude: 1. Does not handle REFER although it claims to do so on the basis of Allow: ( sends BYE right away and no RE-INVITE if I send him the REFER). 2. Does not handle codec renegotiation unless I do SDP answer in ACK before sending a re-invite ( why? ) 3. I have also found bt.com gives you only about 20 seconds of ringing before it hangs up. CONCLUSION: BT.com will only work correctly for all call flows if sipxbridge handles the transfer. To effect this you have to turn OFF MOH on the phone and turn it ON for the bridge. Other configurations will not work for all transfers and besides you will get dead air when the transfer target is ringing. You must also turn OFF RTP Keepalive or your call is dropped when doing the transfer ( media bridge on their end cannot deal with RTP packets it does not recognize ). Ranga -- M. Ranganathan
merged.xml.gz
Description: GNU Zip compressed data
_______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
