Hi, I was searching on SIP Trunking, and found the SIP SOPHIA discussion list (please, see below). I thought that someone here has been trying to meet the recommendations of the SIP connect 1.0. rgs Cesar ========================================================================================================================================================================= More or less, sofia is the only one that both fit my ideal model of a SIP UA *and* can handle the call volume in sip traffic that matches the boxes capabilty every other stack we tried would buckle under the stress and deadlock, crash, run up the CPU or run out of memory. I don't know of any such comparison page, sorry. On Mon, May 19, 2008 at 1:18 PM, Jim Thomas <[EMAIL PROTECTED]> wrote: > Anthony, > > Could you elaborate a little on what led to your selection of Sofia-SIP for > SIP trunking in FreeSwitch? > > What did you like about Sofia-SIP? What didn't you like about the > alternatives? > > Do you know of any online comparison of features, strengths and weaknesses, > for the several SIP stack alternatives? > > Thanks. > > Jim > ----- Original Message ---- > From: Anthony Minessale <[EMAIL PROTECTED]> > To: freeswitch-dev at lists.freeswitch.org > Sent: Thursday, May 15, 2008 9:27:56 AM > Subject: Re: [Freeswitch-dev] Sofia-SIP for SIP trunking > > Well it was the 5th SIP stack we tried including all the other ones on your > list and we have some people doing 400+ calls per second so it's probably a > good choice. > > On Thu, May 15, 2008 at 8:54 AM, Jim Thomas <[EMAIL PROTECTED]> > wrote: > >> Hello, >> >> >> Is Sofia-SIP a good choice to add SIP trunking support to a PBX? >> >> >> Are there any known limitations or weaknesses relative to alternatives >> such as sipX, reSIProcate, etc? >> >> >> Is SIP trunking based on Sofia-SIP interoperable with a broad range of SIP >> trunking service providers? >> >> >> Thanks. >> >> >> Jim >> ============================================================================================================================================================================= > Hi All, > > According to the abstract "IP PBX / Service Provider Interoperability" > ( > http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,134/Itemid,75/ > ) > there is a set of recommendations that must be followed: > > "This document outlines an interface specification that enables direct > IP peering between a SIP-enabled Service Provider network and a > SIP-enabled Enterprise Network for the purpose of originating and/or > terminating calls from the Public Switched Telephone Network (PSTN). > It specifies the minimal set of IETF and ITU-T standards that must be > supported,provides precise guidance in the areas where the standards > leave multiple implementation options, and specifies a minimalset of > capabilities that should be supported by the Service Provider and > Enterprise networks." > > I wonder if the PJSIP agrees with those recommendations, or whether it I think this concerns IP PBX and proxy servers, thus it's not relevant to us. > > is possible to make SIP trunk, using an implementation based on PJSIP. > > Sure. Although I wonder why existing implementations are not suitable for you. As we know there are plenty of open source PBX, SIP proxies, and softswitches out there. Cheers Benny
