asat232

When one sip-phone is making SRTP call to onother sip-phone its sending to pbxnsip INVITE SAVP, but pbxnsip is placing INVITE to other party with AVP. How can i disable this behavior, because if i place INVITE with SAVP then i want INVITE to other party with SAVP too. Hope for your help. Thanks.

I asked you about G729 and SPEEX, because from my point of view their implementations are working well on Windows Mobile platform (with respect to limited computational resources and narrow band). The advantages of speex are comparable quality, and no need to pay for license. Therefore I’ m interested whether its expected to support speex.

So far as it concerned G729 it is clear for me now. Thanks a lot. But what can you tell me about speex codec.

At last I got the idea, needs to buy license and I will get the version of pbxnsip which supported G729, am I right?

You helped me to resolve my problem. Thanks a lot.

Yes, for sure, G729 required licensing, but how it can be enabled. I went as you told me to Ports page and set in codec Preferences for G729 higher priority and again 18 is missing in sdp media description when I making call. audio 62234 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv

I disabled mailbox for both accounts but the same situation first one rejects call on another call esteblished.

Can be disabled this behavior to make it optional?

Is it expected to implement support G729 and SPEEX codecs by pbxnsip? Looking sdp (invite from pbxnsip) i saw only codes 0-pcmu, 8-pcma, 9-g722 2-(i don't remember exactly), 3-GSM.

Our support sometimes is giving such an answer, this is not a bug, and this is a feature. But is it correct behavior when one part is rejecting call and another part is getting 200 ok, and establish call. And pbxnsip considering this call as an active. I think this feature is reproducible and can be easily fixed. What do you think about this matter? Or I somewhere on a wrong way? Thanks in advance for your help.

Here is a logpbxnsip_log.txt

When I dialing from first softphone (EyeBeam 1.5.19.5) to second one (EyeBeam too), and on second one I do HandUp, first EyeBeam is going to call established state. And I can see though web interface that this call is currently active. Looking to pbxnsip logfile page, INVITE response from softphone was 480. Is it a bug, or what is it?