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If I have this is sip.conf :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[general]</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">disallow=all</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">allow=g729</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">allow=alaw</FONT></FONT><BR>
<BR>
The prefered codecs set in my Grandstream phone is G729, alaw.<BR>
<BR>
In the sip peer definition I have commented out 'disallow=' and 'allow='.<BR>
<BR>
The prefered codecs set in the Zoiper softphone is alaw, gsm.<BR>
<BR>
In the sip peer definition I have commented out 'disallow=' and 'allow='.<BR>
<BR>
When making a call from the Grandstream to the Zoiper softphone, with Asterisk staying in the media path (canreinvite=no), you would expect all 3 of them to use alaw.<BR>
But this is what happens :<BR>
<BR>
The Grandstream :<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">v=0</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">o=test3 8000 8001 IN IP4 192.168.1.101</FONT></FONT><FONT COLOR="#000000"> (<-- Grandstream IP-address)</FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">s=SIP Call</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">c=IN IP4 192.168.1.101</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">t=0 0</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">m=audio 10082 RTP/AVP 18 8 101</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">a=sendrecv</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">a=rtpmap:18 G729/8000</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">a=rtpmap:8 PCMA/8000</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">a=ptime:20</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">a=rtpmap:101 telephone-event/8000</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">a=fmtp:101 0-11</FONT></FONT><BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Found RTP audio format 18</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Found RTP audio format 8</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Found RTP audio format 101</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Peer audio RTP is at port 192.168.1.101:10082</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Found audio description format G729 for ID 18</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Found audio description format PCMA for ID 8</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Found audio description format telephone-event for ID 101</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] Peer audio RTP is at port 192.168.1.101:10082</FONT></FONT><BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] -- Called test1 </FONT></FONT><FONT COLOR="#000000">(<-- the Zoiper softphone)</FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Mar 12 11:44:14] -- Got SIP response 415 "Unsupported Media Type" back from 192.168.1.106 </FONT></FONT><FONT COLOR="#000000">(IP-address of Zoiper softphone)</FONT><BR>
<BR>
<BR>
<FONT COLOR="#000000">So what happens here with the codec negotiation between Asterisk and the Zoiper softphone ??</FONT><BR>
<BR>
<FONT COLOR="#000000">Making the call the other way around (Zoiper calls Grandstream) the call succeeds and the codec is alaw... like it should be.</FONT><BR>
<FONT COLOR="#000000">I do get the warnings :</FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[Mar 12 11:53:30] WARNING[23703]: channel.c:3340 ast_channel_make_compatible: No path to translate from SIP/test3-0a168b48(256) to SIP/test1-0a166d00(8)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[Mar 12 11:53:31] -- SIP/test3-0a168b48 is ringing</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[Mar 12 11:53:34] WARNING[23670]: channel.c:2961 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[Mar 12 11:53:34] WARNING[23670]: channel.c:2961 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729)</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[Mar 12 11:53:34] -- SIP/test3-0a168b48 answered SIP/test1-0a166d00</FONT></FONT><BR>
<FONT COLOR="#0000ff"><FONT SIZE="2">[Mar 12 11:53:34] -- Packet2Packet bridging SIP/test1-0a166d00 and SIP/test3-0a168b48</FONT></FONT><BR>
<BR>
(I know there are G729-licences to translate from G729 to alaw, but if both support alaw, then alaw should be the negotiated codec, no ?!)<BR>
<BR>
<FONT COLOR="#000000">Can this be explained ?</FONT><BR>
<BR>
<BR>
<FONT COLOR="#000000">Thanks.</FONT><BR>
<FONT COLOR="#000000">Jonas.</FONT>
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