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I am developing a softphone,using "officesip (https://sourceforge.net/projects/officesip/)", I succeeded in establishing call between subscribers, but while a receiver picks up the call the, the call is terminated,showing "error:488:not acceptable here".

At sender side, I found that the error is related to media compatibility issue. But I couldn't find how to solve it, where do I need to configure or change?

Notes:

  1. I have only headset connected to my PC no webcams

  2. SIP server is NGCP-SIPWISE

  3. The call works perfectly when I call from "officesip" to "X-lite",but terminates when I call between two "officesip",after receiver picks up,with status 488.

Software_developer
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1 Answers1

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The "488 Not acceptable here" means audio codec incompatibility problems. Make sure that both ends has at least one common codec. You can see the codec list in the INVITE message SDP part (with the m=audio line the codec payload numbers are enumerated and also in the a=rtpmap lines as text).

Also it might be a bug in officesip. In this case I would recommend to enable only one single codec on both end (such as PCMU) and try a call.

Istvan
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  • Thank you for your ,help,but i don't know how to check codec which are used on both sides,or to change codec to be used like PCMU,,how can i configure it?and how can i determine codec used on both side? – Software_developer Apr 29 '16 at 11:09
  • I described how to check the codec used. Check my answer: INVITE message SDP parameter. You have to check your softphone logs for this or use Wireshark. – Istvan Apr 29 '16 at 11:17
  • Thank you for your help,here is my log,please help me with that..for sender side: https://www.dropbox.com/s/8zavulg242mnbbj/Sender%28Calling%20person%29.txt?dl=0 and for reciever side: https://www.dropbox.com/s/cjbanhdyoj8abrb/Receiver%28called%20person%29.txt?dl=0 – Software_developer Apr 29 '16 at 11:36
  • This is not enough because i can't see the SIP signaling in the logs. Maybe there is some option to enable also this (higher loglevel) or you should capture a network log with wireshark filtering to "SIP". – Istvan Apr 29 '16 at 14:41
  • ok,thanks again,i have uploded file generated from wireshark in this link https://www.dropbox.com/s/a32mov3a2wv3rql/call-error.pcapng?dl=0 please review it.. – Software_developer Apr 30 '16 at 06:33
  • I see. You should make a test with all encryption disabled in the client, since it is possible that those are not compatible with the server. Also turn off video (So in the INVITE message SDP part should be simplified to audio only) – Istvan Apr 30 '16 at 08:17
  • Let us [continue this discussion in chat](http://chat.stackoverflow.com/rooms/110710/discussion-between-software-developer-and-istvan). – Software_developer Apr 30 '16 at 10:25
  • I am sorry, but I don't have time for chat. – Istvan May 10 '16 at 19:16