# Transcode audio codecs issue pjsip **URL:** \ **Category:** Asterisk Support\ **Created:** [March 12, 2020, 10:44am UTC](https://community.asterisk.org/t/transcode-audio-codecs-issue-pjsip/83074 "2020-03-12T10:44:25Z")\ **Posts on this page:** 6\ **Page:** 1
**Author:** ![sami-hep](https://avatars.discourse-cdn.com/v4/letter/s/ecd19e/32.png) [@sami-hep](https://community.asterisk.org/u/sami-hep)\ **Post date:** [March 12, 2020, 10:44am UTC](https://community.asterisk.org/t/transcode-audio-codecs-issue-pjsip/83074/1 "2020-03-12T10:44:25Z")
at one endpoint i allowed gsm at the other i allowed i allowed alaw and ulaw but the call never happens because of difference of audio codecs although asterisk says that there is transcoding. this is the translation path of gsm: — Translation paths SRC Codec “gsm” sample rate 8000 — gsm:8000 To codec2:8000 : No Translation Path gsm:8000 To g723:8000 : No Translation Path gsm:8000 To ulaw:8000 : (gsm@8000)-\>(slin@8000)-\>(ulaw@8000) gsm:8000 To alaw:8000 : (gsm@8000)-\>(slin@8000)-\>(alaw@8000) gsm:8000 To g726:8000 : (gsm@8000)-\>(slin@8000)-\>(g726@8000) gsm:8000 To g726aal2:8000 : (gsm@8000)-\>(slin@8000)-\>(g726aal2@8000) gsm:8000 To adpcm:8000 : (gsm@8000)-\>(slin@8000)-\>(adpcm@8000) gsm:8000 To slin:8000 : (gsm@8000)-\>(slin@8000) gsm:8000 To slin:12000 : (gsm@8000)-\>(slin@8000)-\>(slin@12000) gsm:8000 To slin:16000 : (gsm@8000)-\>(slin@8000)-\>(slin@16000) gsm:8000 To slin:24000 : (gsm@8000)-\>(slin@8000)-\>(slin@24000) gsm:8000 To slin:32000 : (gsm@8000)-\>(slin@8000)-\>(slin@32000) gsm:8000 To slin:44100 : (gsm@8000)-\>(slin@8000)-\>(slin@44100) gsm:8000 To slin:48000 : (gsm@8000)-\>(slin@8000)-\>(slin@48000) gsm:8000 To slin:96000 : (gsm@8000)-\>(slin@8000)-\>(slin@96000) gsm:8000 To slin:192000 : (gsm@8000)-\>(slin@8000)-\>(slin@192000) gsm:8000 To lpc10:8000 : (gsm@8000)-\>(slin@8000)-\>(lpc10@8000) gsm:8000 To g729:8000 : No Translation Path gsm:8000 To speex:8000 : No Translation Path gsm:8000 To speex:16000 : No Translation Path gsm:8000 To speex:32000 : No Translation Path gsm:8000 To ilbc:8000 : (gsm@8000)-\>(slin@8000)-\>(ilbc@8000) gsm:8000 To g722:16000 : (gsm@8000)-\>(slin@8000)-\>(g722@16000) gsm:8000 To siren7:16000 : No Translation Path gsm:8000 To siren14:32000 : No Translation Path gsm:8000 To testlaw:8000 : (gsm@8000)-\>(slin@8000)-\>(testlaw@8000) gsm:8000 To g719:48000 : No Translation Path gsm:8000 To opus:48000 : No Translation Path gsm:8000 To none:8000 : No Translation Path gsm:8000 To silk:8000 : No Translation Path gsm:8000 To silk:12000 : No Translation Path gsm:8000 To silk:16000 : No Translation Path gsm:8000 To silk:24000 : No Translation Path this is the log: \<— Received SIP request (1079 bytes) from UDP:192.168.133.9:5060 —\> INVITE [sip:37301@192.168.133.111](mailto:sip:37301@192.168.133.111) SIP/2.0 Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.ISnaroGGk;rport From: sip:37200@192.168.133.111;tag=6Sq~N7rxb To: [sip:37301@192.168.133.111](mailto:sip:37301@192.168.133.111) CSeq: 20 INVITE Call-ID: 9t3Bc4p4m7 Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 495 Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=“urn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6” User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3) v=0 o=37200 2645 782 IN IP4 192.168.133.9 s=Talk c=IN IP4 192.168.133.9 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/AVP 0 8 96 97 98 101 99 100 a=rtpmap:96 speex/8000 a=fmtp:96 vbr=on a=rtpmap:97 opus/48000/2 a=fmtp:97 useinbandfec=1 a=rtpmap:98 speex/16000 a=fmtp:98 vbr=on a=rtpmap:101 telephone-event/8000 a=rtpmap:99 telephone-event/48000 a=rtpmap:100 telephone-event/16000 a=rtcp-fb:\* trr-int 5000 a=rtcp-fb:\* ccm tmmbr \<— Transmitting SIP response (475 bytes) to UDP:192.168.133.9:5060 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.ISnaroGGk Call-ID: 9t3Bc4p4m7 From: sip:37200@192.168.133.111;tag=6Sq~N7rxb To: sip:37301@192.168.133.111;tag=z9hG4bK.ISnaroGGk CSeq: 20 INVITE WWW-Authenticate: Digest realm=“asterisk”,nonce=“1584009606/de0be536678b7eaabfff3d900cbe7027”,opaque=“11f0ba1743b05957”,algorithm=md5,qop=“auth” Server: Asterisk PBX certified/16.3-cert1 Content-Length: 0 \<— Received SIP request (380 bytes) from UDP:192.168.133.9:5060 —\> ACK [sip:37301@192.168.133.111](mailto:sip:37301@192.168.133.111) SIP/2.0 Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.ISnaroGGk;rport Call-ID: 9t3Bc4p4m7 From: sip:37200@192.168.133.111;tag=6Sq~N7rxb To: sip:37301@192.168.133.111;tag=z9hG4bK.ISnaroGGk Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=“urn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6” Max-Forwards: 70 CSeq: 20 ACK \<— Received SIP request (1362 bytes) from UDP:192.168.133.9:5060 —\> INVITE [sip:37301@192.168.133.111](mailto:sip:37301@192.168.133.111) SIP/2.0 Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.BXINd8rvn;rport From: sip:37200@192.168.133.111;tag=6Sq~N7rxb To: [sip:37301@192.168.133.111](mailto:sip:37301@192.168.133.111) CSeq: 21 INVITE Call-ID: 9t3Bc4p4m7 Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 495 Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=“urn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6” User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm=“asterisk”, nonce=“1584009606/de0be536678b7eaabfff3d900cbe7027”, algorithm=md5, opaque=“11f0ba1743b05957”, username=“37200”, [uri="sip:37301@192.168.133.111](mailto:uri=%22sip:37301@192.168.133.111)", response=“79bdff45377b73e674c88d185961d636”, cnonce=“FpoXmJMmZIWIabfS”, nc=00000001, qop=auth v=0 o=37200 2645 782 IN IP4 192.168.133.9 s=Talk c=IN IP4 192.168.133.9 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/AVP 0 8 96 97 98 101 99 100 a=rtpmap:96 speex/8000 a=fmtp:96 vbr=on a=rtpmap:97 opus/48000/2 a=fmtp:97 useinbandfec=1 a=rtpmap:98 speex/16000 a=fmtp:98 vbr=on a=rtpmap:101 telephone-event/8000 a=rtpmap:99 telephone-event/48000 a=rtpmap:100 telephone-event/16000 a=rtcp-fb:\* trr-int 5000 a=rtcp-fb:\* ccm tmmbr == Setting global variable ‘SIPDOMAIN’ to ‘192.168.133.111’ \<— Transmitting SIP response (301 bytes) to UDP:192.168.133.9:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.BXINd8rvn Call-ID: 9t3Bc4p4m7 From: sip:37200@192.168.133.111;tag=6Sq~N7rxb To: sip:37301@192.168.133.111 CSeq: 21 INVITE Server: Asterisk PBX certified/16.3-cert1 Content-Length: 0 **[Mar 12 12:40:06] NOTICE[16156]: res\_pjsip\_sdp\_rtp.c:418 set\_caps: No joint capabilities for ‘audio’ media stream between our configuration((gsm)) and incoming SDP((ulaw|alaw|speex|opus|speex16))** \<— Transmitting SIP response (355 bytes) to UDP:192.168.133.9:5060 —\> SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.BXINd8rvn Call-ID: 9t3Bc4p4m7 From: sip:37200@192.168.133.111;tag=6Sq~N7rxb To: sip:37301@192.168.133.111;tag=695f8456-d5d4-4e18-b81b-3c64b9f8828c CSeq: 21 INVITE Server: Asterisk PBX certified/16.3-cert1 Content-Length: 0 \<— Received SIP request (399 bytes) from UDP:192.168.133.9:5060 —\> ACK [sip:37301@192.168.133.111](mailto:sip:37301@192.168.133.111) SIP/2.0 Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.BXINd8rvn;rport Call-ID: 9t3Bc4p4m7 From: sip:37200@192.168.133.111;tag=6Sq~N7rxb To: sip:37301@192.168.133.111;tag=695f8456-d5d4-4e18-b81b-3c64b9f8828c Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=“urn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6” Max-Forwards: 70 CSeq: 21 ACK ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [March 12, 2020, 11:08am UTC](https://community.asterisk.org/t/transcode-audio-codecs-issue-pjsip/83074/2 "2020-03-12T11:08:19Z")
The call between an endpoint and Asterisk has to agree on at least one common codec to work. You can’t send gsm to an endpoint if it was never negotiated. Transcoding occurs when you are playing back files, or talking to another channel with a different codec. ---
**Author:** ![sami-hep](https://avatars.discourse-cdn.com/v4/letter/s/ecd19e/32.png) [@sami-hep](https://community.asterisk.org/u/sami-hep)\ **Post date:** [March 12, 2020, 11:09am UTC](https://community.asterisk.org/t/transcode-audio-codecs-issue-pjsip/83074/3 "2020-03-12T11:09:17Z")
what do you mean by to another channel ? ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [March 12, 2020, 11:10am UTC](https://community.asterisk.org/t/transcode-audio-codecs-issue-pjsip/83074/4 "2020-03-12T11:10:00Z")
If endpoint Alice calls endpoint Bob through Asterisk, then that would be two channels. Alice could have negotiated as ulaw, and Bob as gsm. In that case Asterisk would transcode between ulaw and gsm. ---
**Author:** ![sami-hep](https://avatars.discourse-cdn.com/v4/letter/s/ecd19e/32.png) [@sami-hep](https://community.asterisk.org/u/sami-hep)\ **Post date:** [March 12, 2020, 11:21am UTC](https://community.asterisk.org/t/transcode-audio-codecs-issue-pjsip/83074/5 "2020-03-12T11:21:19Z")
got it. that was so helpful ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 11, 2020, 11:21am UTC](https://community.asterisk.org/t/transcode-audio-codecs-issue-pjsip/83074/6 "2020-04-11T11:21:25Z")
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