Hi, I have compiled a pjsip version with APS (Audio Proxy Server). By default the used codec is PCMU. How can I change to use iLBC? I followed the wiki indications section "Using APS-Direct and VAS-Direct in PJMEDIA" which suggests to enable the passthrough codecs: 1. Enable passthrough codecs, and selectively enable/disable which passthrough codecs to be supported. The passthrough codecs supported would depend on which codecs are supported by the sound device backend that you choose to use: 2. #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 3. 4. // Disable all passthrough codecs except PCMA and PCMU 5. #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 0 6. #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 0 7. #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0 8. #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 0 9. #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 1 I set PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC = 1. I activated the log and in the log I got: ******* Opening sound device PCM at 8000/1/20msPort 1 (sip:5148403000 at sip6.van.netvoice.ca) transmitting to port 0 (S60 APS)Port 0 (S60 APS) transmitting to port 1 (sip:5148403000 at sip6.van.netvoice.ca)Player initialized, err=0Processing incoming message: Response msg 200/INVITE/cseq=18842 (rdata0x71966c)RX 855 bytes Response msg 200/INVITE/cseq=18842 (rdata0x71966c) from UDP 64.34.49.82:5060: SIP/2.0 200 OK ************ My SIP server responds to INVITE with the following media parameters: --end msg--Incoming Response msg 100/INVITE/cseq=18842 (rdata0x71966c) in state CallingState changed from Calling to Proceeding, event=RX_MSGReceived Response msg 100/INVITE/cseq=18842 (rdata0x71966c)Transaction tsx0x73447c state changed to ProceedingProcessing incoming message: Response msg 183/INVITE/cseq=18842 (rdata0x71966c)RX 869 bytes Response msg 183/INVITE/cseq=18842 (rdata0x71966c) from UDP 64.34.49.82:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKPj3NCPl88fFmKGLCHBUcWBn0g5K7a.gBTw;received =67.226.182.65;rport=21718 From: sip:nv2.ctci01.a at sip6.van.netvoice.ca;tag=k1EszvvqimbJFTviUnhnOyBnUgYMM3N1 To: sip:5148403000 at sip6.van.netvoice.ca;tag=as2a2181f5 Call-ID: q71cnw0uIE5OzvGJnlaeb-hnQmt2s9-T CSeq: 18842 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:5148403000 at 64.34.49.82> Content-Type: application/sdp Content-Length: 309 v=0 o=root 26511 26511 IN IP4 64.34.49.82 s=session c=IN IP4 64.34.49.82 t=0 0 m=audio 11430 RTP/AVP 0 3 113 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:113 iLBC/8000 a=fmtp:113 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 We can see the 1st codec is PCMU/8000. Could someone give me some suggestions? Thank you, George. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090604/d17484b1/attachment-0001.html>
