How can I see which codec is in use during conversation. I can see (for example) which codecs are negotiated before SIP connection, but I don't know which is chosen: 12 headers, 12 lines Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 217.10.79.30:15666 Found description format GSM Found description format iLBC Found description format G726-32 Found description format telephone-event Capabilities: us - 0x693(G723|GSM|G726|LPC10|SPEEX|ILBC), peer - audio=0x412(GSM|G726|ILBC)/video=0x0(EMPTY), combined - 0x412(GSM|G726|ILBC) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) set_destination: Parsing <sip:10000@217.10.79.9;ftag=as61888e0f;lr=on> for address/port to send to set_destination: set destination to 217.10.79.9, port 5060 Transmitting: ACK sip:10000@sipgate.de SIP/2.0 Via: SIP/2.0/UDP 213.240.3.178:5060;branch=z9hG4bK7c84b82f Route: <sip:10000@217.10.79.30> From: "Test" <sip:xxxxxxx@sipgate.de>;tag=as61888e0f To: <sip:10000@sipgate.de>;tag=as2fb68341 Contact: <sip:xxxxxxx@213.240.3.178> Call-ID: 1f85a3e41713ae3273aca539042390b1@213.240.3.178 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0