Transcode audio codecs issue pjsip

Asterisk Asterisk Support
1

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 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
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 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 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
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", 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 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

2

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.

1 Like
3

what do you mean by to another channel ?

4

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.

1 Like
5

got it. that was so helpful

6

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.