sip "100 trying" instead of "180 ringing"
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.
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Yuriy Vikulov almost 2 years
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:[email protected]:45134;rinstance=7af05ded7e7e49e6> To: <sip:[email protected]:45134;rinstance=7af05ded7e7e49e6>;tag=9a00d038 From: "6012"<sip:[email protected]>;tag=as66995cd4 Call-ID: [email protected] 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:[email protected]>;tag=18345648 To: "6012"<sip:[email protected]> 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:[email protected]> 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?