Hi There, I'm looking to build a simple auto-answering SIP client using PJSUA that runs on a raspberry pi. Right now I'm running into a few problems that I'm hoping I might be able to further understand and fix: * PJSUA seems to correctly start the call by auto-answering and returning call code 200, but no audio is transferred between the clients. * After 30 seconds the call gets terminated. I don't understand why it terminates or whether this termination was triggered by the client or server. * After this termination, the console displays ">>>>" instead of ">>>" and I'm unable to input any further commands. Is this because of some error in ending the call? For some context, I'm making calls between extensions on a local asterisk server. This asterisk server is running in virtualbox on a separate mac. Besides this client, I'm using two others, one iphone based one using an app called Media5-fone, and another on a windows computer running a GUI based SIP client called ekiga. When I get these two clients to call each other, they do so without any issues, so I think the problem is isolated to PJSUA. I've put up the trace level logs here:?https://gist.github.com/zachgoldstein/5807176 [1]. The config file I'm using when starting PJSUA is here:?https://gist.github.com/zachgoldstein/5807663 [2] Initially I thought it was an issue with the usb headset I'm using, thinking that it's not properly setup. However, using the cl command I can see that the headset shows up during a call and connects to what I think is the proper port:>>> clConference ports:Port #00[44KHz/20ms/1] DYNEX USB Audio Device: USB Audio (hw:0,0) ?transmitting to: #3?Port #01[44KHz/20ms/1] ? ? ? ? ? ? ringback ?transmitting to:?Port #02[44KHz/20ms/1] ? ? ? ? ? ? ? ? ring ?transmitting to:?Port #03[ 8KHz/20ms/1] ?sip:2 at 192.168.1.119 [3]??transmitting to: #0? Additionally, when I run PJsystest, I'm able to hear the tone when using test 02. (Other two wav file tests cannot find files). I take this to mean that I should at least hear audio from the other client's microphone during calls. So I then thought that it might be because of excessive CPU load while the call is occurring since the Raspberry Pis aren't exactly the beefiest machines. So I put --ec-tail 0 into my config file, hoping that might be enough. Running top at the same time as PJSUA doesn't show that it's consuming large amounts of CPU or memory, but I notice a massive lag with the mouse when I'm in the Raspberry Pi GUI. All bash instances also become unresponsive, forcing me to end the process. To get around this, I ssh into the machine, where the lag seems to not exist. I think this could be a NAT issue.?This page [4]?outlines that the output of using dq should show non-zero values for RX pt. Mine shows as 0 despite the total pkts being of some reasonable value. None of the three options shown on?this page [5]?worked to resolve the issue. Right now I'm plugged into a router that bridges into another router, could that cause problems with NAT? I've run out of ideas for what could be wrong here. I'm really sorry to bother you guys about this. Any help is massively appreciated! Thanks,-Zach Links: ------ [1] https://gist.github.com/zachgoldstein/5807176 [2] https://gist.github.com/zachgoldstein/5807663 [3] mailto:sip%3A2 at 192.168.1119 [4] http://trac.pjsip.org/repos/wiki/audio-check-rx-rtp [5] http://trac.pjsip.org/repos/wiki/Audio_Problems/Getting_Around_Nat -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130621/3753da41/attachment-0001.html>
