sip “100 trying” instead of “180 ringing”

Question

I develop a software using Microsoft Unified Communications and c#. I'm using a IMVoipSample as a code base. As a voip backend i'm using asterisk. Everything fine, i can register, make calls, accept/reject incoming calls. There is a one thing that i cannot resolve.

while i make a call to a 3rd party softphone there is an answer from it:

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.250.29:5060;branch=z9hG4bK786d156c;rport=5060
Contact: <sip:6011@192.168.246.203:45134;rinstance=7af05ded7e7e49e6>
To: <sip:6011@192.168.246.203:45134;rinstance=7af05ded7e7e49e6>;tag=9a00d038
From: "6012"<sip:6012@192.168.250.29>;tag=as66995cd4
Call-ID: 7cebe5d1060b11452571a22e0e2cb919@192.168.250.29
CSeq: 102 INVITE
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 0

But when i make a call to my IMVoipSample phone there is an aswer:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.246.203:45134;branch=z9hG4bK-d87543-71570d1c6127bc7a-1--d87543-;received=192.168.246.203;rport=45134
From: "6011"<sip:6011@192.168.250.29>;tag=18345648
To: "6012"<sip:6012@192.168.250.29>
Call-ID: fd7f305d6520cd53YjQ2ZDJmMDAwZDE0YmUwMjRlMGFmM2NmODg5OGM1ODQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:6012@192.168.250.29>
Content-Length: 0

I have a samsung officeserv pbx, it is connected to asterisk, i can make calls to softphones and vice verca. But the audio difference of making calls to softphone and my IMVoipSample phone is there is no normal connecting beeps, only silence. I suppose it is because of "sip 100 trying" instead of "180 rinning". So the question is: do I need setup additional signalling of ringing in client?

Answer 1

1xx responses are informational responses and in the case of 100 Trying are optional. SIP User Agent Servers (UAS) will generally respond with a 100 Trying response immediately when they receive an INVITE request to let the User Agent Client (UAC) know they are processing the request and to avoid retransmits. At some time later they will follow the 100 Trying response with a 180 Ringing or 183 Session Progress. Once someone or something answers the call a 2xx response needs to be sent, typically 200 Ok.

If your softphone software is only generating a 100 Trying response and not the subsequent 180 Ringing response then my guess is you have missed a step. At the very least if the softphone has a problem and can't generate a ringing response because there is nothing to ring it should generate a 4xx error response.

Answer 2

Well, i solved my issue by explitic adding ringing in sip.conf of asterisk

exten => _6.,1,Ringing
exten => _6.,n,Dial(SIP/${EXTEN:0}).

but @sipwiz gave a detailed description of the source of problem