Log comparing 1.0.1 and 1.0-RC2 if a call from a local phone (GSIn) via Asterisk is made: The phone first calls Asterisk, in both releases Asterisk "only" has GSM/ULAW/ALAW/H263: Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Both then call the provider (Sipgate) and negotiate to ilbc (no auth needed as we call a tollfree number): Reliably Transmitting: INVITE sip:08003301000 at sipgate.de SIP/2.0 Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK4790c355 From: "1838074" ;tag=as33f18800 To: Contact: Call-ID: 1de5c3cc4a81df5c56aa0dcb0b090ea7 at 217.162.36.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 30 Sep 2004 17:32:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 162 v=0 o=root 4571 4571 IN IP4 217.162.36.14 s=session c=IN IP4 217.162.36.14 t=0 0 m=audio 14968 RTP/AVP 97 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - ... Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK4790c355 From: "1838074" ;tag=as33f18800 To: ;tag=as2cad2e8e Call-ID: 1de5c3cc4a81df5c56aa0dcb0b090ea7 at 217.162.36.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 160 v=0 o=root 1179 1179 IN IP4 217.10.79.30 s=session c=IN IP4 217.10.79.30 t=0 0 m=audio 17846 RTP/AVP 97 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - ... Found description format iLBC Capabilities: us - 0x400(ILBC), peer - audio=0x400(ILBC)/video=0x0(EMPTY), combined - 0x400(ILBC) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) -- SIP/Sipgate-da7c is making progress passing it to SIP/GSIn-cc2f Both answer back to the phone device, initially still with GSM/ULAW/ALAW: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK14147804d4d8d453 From: ;tag=0ec4d9401783a949 To: ;tag=as30e3d88b Call-ID: 6e0445ae3e8e3589 at 10.1.41.188 CSeq: 53443 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 197 v=0 o=root 4571 4571 IN IP4 10.1.1.1 s=session c=IN IP4 10.1.1.1 t=0 0 m=audio 12436 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - Now the callee picks up the phone, and Sipgate sends an OK. Now it comes... So far, both releases behaved similar, but here they differ, as soon as the SIP_CODEC variable was changed in extensions.conf (both perform this, as the note below shows): This happens in 1.0 RC2: Sep 30 19:32:41 NOTICE[294928]: chan_sip.c:1817 sip_answer: Changing codec to 'ilbc' for this call because of ${SIP_CODEC) variable We're at 10.1.1.1 port 12436 Answering with capability 0x400(ILBC) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK14147804d4d8d453 From: ;tag=0ec4d9401783a949 To: ;tag=as30e3d88b Call-ID: 6e0445ae3e8e3589 at 10.1.41.188 CSeq: 53443 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 152 v=0 o=root 4571 4572 IN IP4 10.1.1.1 s=session c=IN IP4 10.1.1.1 t=0 0 m=audio 12436 RTP/AVP 99 a=rtpmap:99 iLBC/8000 <<<<<<<<<<<<<<<<< That's ok!!!! a=silenceSupp:off - - - - to 10.1.41.188:5060 -- Attempting native bridge of SIP/GSIn-cc2f and SIP/Sipgate-da7c -- Attempting native bridge of SIP/GSIn-cc2f and SIP/Sipgate-da7c Opposed to it, this happens in 1.0.1: -- SIP/Sipgate-a0b3 answered SIP/GSIn-285d Sep 30 19:14:10 NOTICE[245775]: chan_sip.c:1853 sip_answer: Changing codec to 'ilbc' for this call because of ${SIP_CODEC) variable We're at 10.1.1.1 port 19998 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK508e666cda524530 From: ;tag=bec21ba9e2c638f5 To: ;tag=as1e1c9d3f Call-ID: 26ca9bc8cd4a4982 at 10.1.41.188 CSeq: 53206 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 197 v=0 o=root 2744 2745 IN IP4 10.1.1.1 s=session c=IN IP4 10.1.1.1 t=0 0 m=audio 19998 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 <<<<<<<<<<<<<<<<< should be ilbc!!!! a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - to 10.1.41.188:5060 -- Attempting native bridge of SIP/GSIn-285d and SIP/Sipgate-a0b3 Consequently, "sip show channels" in 1.0 RC2: rufus*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 217.10.79.9 0800330100 1de5c3cc4a8 00102/00000 ILBC 10.1.41.188 GSIn 6e0445ae3e8 00101/53443 ILBC 2 active SIP channel(s) and in 1.0.1: rufus*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 217.10.79.9 0800330100 4dceed4d2f1 00102/00000 ILBC 10.1.41.188 GSIn 26ca9bc8cd4 00101/53206 ULAW 2 active SIP channel(s)