Asterisk 15.0 + sipML5 -> Strange audio problem


#1

Hi everybody,

after a lot of trail and error, and managed to make calls from a sipML5 (latest release) enabled Chrome Browser to another SIP client (my iPhone). The VoIP stuff is managed by an Asterisk 15.0 server.
Both clients can send and receive audio data. To point it again: These calls are initiated by the sipML5/WebRTC client.

If I reverse it (starting a call from the SIP client), my strange problem occurs:
The sipML5/WebRTC client sends audio data to the SIP client, where I also can hear it. But no audio could be received by the browser. I can’t find any error message in the Asterisk log or the browsers console log.

By enabeling the rtp debugging in the Asterisk CLI, I can see that data is sent from the iPhone (SIP client) to the browser, but no sound comes out of the speakers.

Does anybody have already been faced with a similar problem? Searching the web I found a lot of issues where even no audio is transmitted, but that’s not my case.

Everything is running in my local LAN environment.

Attached you will find my current Asterisk configuration:

sip.conf

[general]
udpbindaddr=0.0.0.0:5060
realm=192.168.0.41
transport=udp,ws,wss
nat=force_rport,comedia

[199] ;WebRTC client
host=dynamic
secret=199 
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=opus
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
rtcp_mux=yes
nat=force_rport,comedia

[201];SIP client (iPhone)
type=friend
username=201
host=dynamic
secret=201
context=default
nat=force_rport,comedia

http.conf

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.pem

rtp.conf

[general]
rtpstart=19000
rtpend=21000
icesupport=true
stunaddr=stun.l.google.com:19302

#2

Please post a sip debug trace (“sip set debug”) of the call in question. Also it should work either way, but you might want to try configuring a pjsip endpoint to see if it’s a chan_sip problem vs pjsip.


#3

Hello kharwell,

Thanks for your answer.

Here’s the debug log your were asking for:

Calling from SIP client (192.168.0.16) to WebRTC client (192.168.0.25)

<--- SIP read from UDP:192.168.0.16:50713 --->
INVITE sip:199@192.168.0.41 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.16:50713;branch=z9hG4bK.4ys9fhq4y;rport
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>
CSeq: 20 INVITE
Call-ID: umaGq8fsNV
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 1935
Contact: <sip:201@192.168.0.16:50713;transport=udp>;+sip.instance="<urn:uuid:11146d96-809d-4751-bc8e-80ae8cd5c3a1>"
User-Agent: Linphone_iPhone5.2_iOS10.3.3/3.16.3 (belle-sip/1.6.1)

v=0
o=201 616 410 IN IP4 192.168.0.16
s=Talk
c=IN IP4 192.168.0.16
b=AS:380
t=0 0
a=ice-pwd:9bd81ae2829c49a261d8329c
a=ice-ufrag:4878b92c
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7272 RTP/AVP 96 97 98 99 0 8 3 9 100 18 102 103 104 105 106 107 108 101 109 110 111 112 113 114
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 SILK/16000
a=rtpmap:98 speex/16000
a=fmtp:98 vbr=on
a=rtpmap:99 speex/8000
a=fmtp:99 vbr=on
a=rtpmap:100 iLBC/8000
a=fmtp:100 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:102 mpeg4-generic/16000
a=fmtp:102 config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:103 mpeg4-generic/22050
a=fmtp:103 config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:104 mpeg4-generic/32000
a=fmtp:104 config=F8E82000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:105 mpeg4-generic/44100
a=fmtp:105 config=F8E82000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:106 mpeg4-generic/48000
a=fmtp:106 config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:107 iSAC/16000
a=rtpmap:108 SILK/24000
a=rtpmap:101 telephone-event/48000
a=rtpmap:109 telephone-event/16000
a=rtpmap:110 telephone-event/8000
a=rtpmap:111 telephone-event/22050
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/44100
a=rtpmap:114 telephone-event/24000
a=candidate:1 1 UDP 2130706431 192.168.0.16 7272 typ host
a=candidate:1 2 UDP 2130706430 192.168.0.16 7273 typ host
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (13 headers 43 lines) ---
Sending to 192.168.0.16:50713 (NAT)
Sending to 192.168.0.16:50713 (NAT)
Using INVITE request as basis request - umaGq8fsNV
Found peer '201' for '201' from 192.168.0.16:50713

<--- Reliably Transmitting (NAT) to 192.168.0.16:50713 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.16:50713;branch=z9hG4bK.4ys9fhq4y;received=192.168.0.16;rport=50713
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>;tag=as2cf152cc
Call-ID: umaGq8fsNV
CSeq: 20 INVITE
Server: Asterisk PBX 15.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.0.41", nonce="61105bac"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'umaGq8fsNV' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.16:50713 --->
ACK sip:199@192.168.0.41 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.16:50713;branch=z9hG4bK.4ys9fhq4y;rport
Call-ID: umaGq8fsNV
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>;tag=as2cf152cc
Contact: <sip:201@192.168.0.16:50713;transport=udp>;+sip.instance="<urn:uuid:11146d96-809d-4751-bc8e-80ae8cd5c3a1>"
Max-Forwards: 70
CSeq: 20 ACK

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.16:50713 --->
INVITE sip:199@192.168.0.41 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.16:50713;branch=z9hG4bK.MflJs8Xbb;rport
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>
CSeq: 21 INVITE
Call-ID: umaGq8fsNV
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 1935
Contact: <sip:201@192.168.0.16:50713;transport=udp>;+sip.instance="<urn:uuid:11146d96-809d-4751-bc8e-80ae8cd5c3a1>"
User-Agent: Linphone_iPhone5.2_iOS10.3.3/3.16.3 (belle-sip/1.6.1)
Authorization: Digest realm="192.168.0.41", nonce="61105bac", algorithm=MD5, username="201", uri="sip:199@192.168.0.41", response="823bf04947a076f989e3c9bb8e743072"

v=0
o=201 616 410 IN IP4 192.168.0.16
s=Talk
c=IN IP4 192.168.0.16
b=AS:380
t=0 0
a=ice-pwd:9bd81ae2829c49a261d8329c
a=ice-ufrag:4878b92c
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7272 RTP/AVP 96 97 98 99 0 8 3 9 100 18 102 103 104 105 106 107 108 101 109 110 111 112 113 114
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 SILK/16000
a=rtpmap:98 speex/16000
a=fmtp:98 vbr=on
a=rtpmap:99 speex/8000
a=fmtp:99 vbr=on
a=rtpmap:100 iLBC/8000
a=fmtp:100 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:102 mpeg4-generic/16000
a=fmtp:102 config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:103 mpeg4-generic/22050
a=fmtp:103 config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:104 mpeg4-generic/32000
a=fmtp:104 config=F8E82000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:105 mpeg4-generic/44100
a=fmtp:105 config=F8E82000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:106 mpeg4-generic/48000
a=fmtp:106 config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5
a=rtpmap:107 iSAC/16000
a=rtpmap:108 SILK/24000
a=rtpmap:101 telephone-event/48000
a=rtpmap:109 telephone-event/16000
a=rtpmap:110 telephone-event/8000
a=rtpmap:111 telephone-event/22050
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/44100
a=rtpmap:114 telephone-event/24000
a=candidate:1 1 UDP 2130706431 192.168.0.16 7272 typ host
a=candidate:1 2 UDP 2130706430 192.168.0.16 7273 typ host
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (14 headers 43 lines) ---
Sending to 192.168.0.16:50713 (NAT)
Using INVITE request as basis request - umaGq8fsNV
Found peer '201' for '201' from 192.168.0.16:50713
  == Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 100
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 106
Found RTP audio format 107
Found RTP audio format 108
Found RTP audio format 101
Found RTP audio format 109
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 114
Found audio description format opus for ID 96
Found unknown media description format SILK for ID 97
Found audio description format speex for ID 98
Found audio description format speex for ID 99
Found audio description format iLBC for ID 100
Found unknown media description format mpeg4-generic for ID 102
Found unknown media description format mpeg4-generic for ID 103
Found unknown media description format mpeg4-generic for ID 104
Found unknown media description format mpeg4-generic for ID 105
Found unknown media description format mpeg4-generic for ID 106
Found unknown media description format iSAC for ID 107
Found unknown media description format SILK for ID 108
Found unknown media description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 109
Found audio description format telephone-event for ID 110
Found unknown media description format telephone-event for ID 111
Found unknown media description format telephone-event for ID 112
Found unknown media description format telephone-event for ID 113
Found unknown media description format telephone-event for ID 114
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|g729|slin192|speex16|h264|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.16:7272
Looking for 199 in default (domain 192.168.0.41)
sip_route_dump: route/path hop: <sip:201@192.168.0.16:50713;transport=udp>

<--- Transmitting (NAT) to 192.168.0.16:50713 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.16:50713;branch=z9hG4bK.MflJs8Xbb;received=192.168.0.16;rport=50713
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>
Call-ID: umaGq8fsNV
CSeq: 21 INVITE
Server: Asterisk PBX 15.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:199@192.168.0.41:5060>
Content-Length: 0


<------------>
    -- Executing [199@default:1] Dial("SIP/201-00000022", "SIP/199") in new stack
  == DTLS ECDH initialized (secp256r1), faster PFS enabled
  == Using SIP RTP CoS mark 5
Audio is at 20866
Adding codec ulaw to SDP
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.25:51573:
INVITE sip:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.0.41:5060;branch=z9hG4bK1bc55d5b;rport
Max-Forwards: 70
From: "test" <sip:201@192.168.0.41>;tag=as733f72cd
To: <sip:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Contact: <sip:201@192.168.0.41:5060;transport=ws>
Call-ID: 3671ee8555233dc62ca3ba5962dc1458@192.168.0.41:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.1.0
Date: Tue, 14 Nov 2017 07:23:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 867

v=0
o=root 719813356 719813356 IN IP4 192.168.0.41
s=Asterisk PBX 15.1.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 20866 RTP/SAVPF 0 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=ice-ufrag:77cbf38b3c713716532908011bdda461
a=ice-pwd:17a3fc3f405c1d5d1431b9af3b451632
a=candidate:Hc0a80029 1 UDP 2130706431 192.168.0.41 20866 typ host
a=candidate:Sbcc2d2a5 1 UDP 1694498815 <my public ip address> 20866 typ srflx raddr 192.168.0.41 rport 20866
a=candidate:Hc0a80029 2 UDP 2130706430 192.168.0.41 20867 typ host
a=candidate:Sbcc2d2a5 2 UDP 1694498814 <my public ip address> 20867 typ srflx raddr 192.168.0.41 rport 20867
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 F4:BA:B9:3B:C9:47:6A:B9:94:23:FD:B3:02:2D:AF:65:24:10:F5:70:F8:94:1C:72:DD:EF:B2:DC:2F:8C:11:B5
a=rtcp-mux
a=sendrecv

---
    -- Called SIP/199

<--- SIP read from WS:192.168.0.25:51573 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 192.168.0.41:5060;rport=5060;branch=z9hG4bK1bc55d5b
From: "test"<sip:201@192.168.0.41>;tag=as733f72cd
To: <sip:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
Call-ID: 3671ee8555233dc62ca3ba5962dc1458@192.168.0.41:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from WS:192.168.0.25:51573 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.0.41:5060;rport=5060;branch=z9hG4bK1bc55d5b
From: "test"<sip:201@192.168.0.41>;tag=as733f72cd
To: <sip:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=ku6HCu5Asx6zvCCTGLd7
Contact: <sip:199@df7jal23ls0d.invalid;transport=ws>
Call-ID: 3671ee8555233dc62ca3ba5962dc1458@192.168.0.41:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:199@df7jal23ls0d.invalid;transport=ws>
    -- SIP/199-00000023 is ringing

<--- Transmitting (NAT) to 192.168.0.16:50713 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.16:50713;branch=z9hG4bK.MflJs8Xbb;received=192.168.0.16;rport=50713
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>;tag=as0c879a79
Call-ID: umaGq8fsNV
CSeq: 21 INVITE
Server: Asterisk PBX 15.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:199@192.168.0.41:5060>
Content-Length: 0


<------------>
    -- SIP/199-00000023 requested media update control 26, passing it to SIP/201-00000022

<--- SIP read from WS:192.168.0.25:51573 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.0.41:5060;rport=5060;branch=z9hG4bK1bc55d5b
From: "test"<sip:201@192.168.0.41>;tag=as733f72cd
To: <sip:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=ku6HCu5Asx6zvCCTGLd7
Contact: <sip:199@df7jal23ls0d.invalid;transport=ws>
Call-ID: 3671ee8555233dc62ca3ba5962dc1458@192.168.0.41:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 945
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=- 928338665493663500 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=msid-semantic: WMS 3zKuRxPQF0PzZGRkyrd4HSp4eKbwXAhqTqED
m=audio 60643 UDP/TLS/RTP/SAVPF 0 107 101
c=IN IP4 192.168.0.25
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:620969918 1 udp 2113937151 192.168.0.25 60643 typ host generation 0 network-cost 50
a=ice-ufrag:oAGB
a=ice-pwd:oFhSKNv99lTefwKZrNLl3VGN
a=ice-options:trickle
a=fingerprint:sha-256 0D:96:27:B1:BD:44:A7:BD:43:1D:D9:C2:FE:D6:8D:77:F2:40:AE:6A:AA:98:05:58:3B:7E:86:56:26:57:A7:D6
a=setup:active
a=mid:audio
a=sendrecv
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=maxptime:20
a=ssrc:2304629353 cname:GWQcpLoebuTqKS/O
a=ssrc:2304629353 msid:3zKuRxPQF0PzZGRkyrd4HSp4eKbwXAhqTqED 3b6fda6a-a329-4106-ab0b-16f8a25b6243
a=ssrc:2304629353 mslabel:3zKuRxPQF0PzZGRkyrd4HSp4eKbwXAhqTqED
a=ssrc:2304629353 label:3b6fda6a-a329-4106-ab0b-16f8a25b6243
<------------->
--- (10 headers 25 lines) ---
Found RTP audio format 0
Found RTP audio format 107
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format opus for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - (opus|ulaw), peer - audio=(ulaw|opus)/video=(nothing)/text=(nothing), combined - (opus|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.25:60643
sip_route_dump: route/path hop: <sip:199@df7jal23ls0d.invalid;transport=ws>
Transmitting (NAT) to 192.168.0.25:51573:
ACK sip:199@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.0.41:5060;branch=z9hG4bK5e2dce29;rport
Max-Forwards: 70
From: "test" <sip:201@192.168.0.41>;tag=as733f72cd
To: <sip:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=ku6HCu5Asx6zvCCTGLd7
Contact: <sip:201@192.168.0.41:5060;transport=ws>
Call-ID: 3671ee8555233dc62ca3ba5962dc1458@192.168.0.41:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.1.0
Content-Length: 0


---
    -- SIP/199-00000023 answered SIP/201-00000022
Audio is at 20838
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.0.16:50713 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.16:50713;branch=z9hG4bK.MflJs8Xbb;received=192.168.0.16;rport=50713
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>;tag=as0c879a79
Call-ID: umaGq8fsNV
CSeq: 21 INVITE
Server: Asterisk PBX 15.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:199@192.168.0.41:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1132442393 1132442393 IN IP4 192.168.0.41
s=Asterisk PBX 15.1.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 20838 RTP/AVP 0 8 3 110
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/199-00000023 joined 'simple_bridge' basic-bridge <f9a22b0d-0648-44e1-ac47-9262f0c6ec00>
    -- Channel SIP/201-00000022 joined 'simple_bridge' basic-bridge <f9a22b0d-0648-44e1-ac47-9262f0c6ec00>

<--- SIP read from UDP:192.168.0.16:50713 --->
ACK sip:199@192.168.0.41:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.16:50713;rport;branch=z9hG4bK.Bm6iNuoiF
From: "test" <sip:201@192.168.0.41>;tag=fw9da609d
To: "199" <sip:199@192.168.0.41>;tag=as0c879a79
CSeq: 21 ACK
Call-ID: umaGq8fsNV
Max-Forwards: 70
Authorization: Digest realm="192.168.0.41", nonce="61105bac", algorithm=MD5, username="201", uri="sip:199@192.168.0.41", response="823bf04947a076f989e3c9bb8e743072"
User-Agent: Linphone_iPhone5.2_iOS10.3.3/3.16.3 (belle-sip/1.6.1)

I’ll will also follow your advice and try to use pjsip… will get back when I have news…

Thanks in advance for your help!

Regards,

fuppy


#5

Unfortunately nothing really sticks out as to the problem. Try limiting your codecs on the endpoints to a single same type to rule out a translation/codec selection problem. For instance for both endpoint configurations set something like the following (allow only one codec):

disallow=all
allow=ulaw