[asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

Richard Mudgett rmudgett at digium.com
Thu May 14 10:01:02 CDT 2020 

Argh.  That was for chan_pjsip and you are using chan_sip.  Be aware that
chan_sip is effectively dead.

Richard

On Thu, May 14, 2020 at 9:50 AM Richard Mudgett <rmudgett at digium.com> wrote:

> The other end is sending g729 even though it was not negotiated.  The
> other end should not do this and it usually seems that the other ends that
> do send g729.
> This was recently fixed.  See
> https://issues.asterisk.org/jira/browse/ASTERISK-28139
>
> Richard
>
> On Thu, May 14, 2020 at 1:11 AM John Hughes <john at calva.com> wrote:
>
>> I am having a problem with one of my callers who is using either g729 or
>> alaw.  I can do alaw but not g729 so asterisk should negotiate alaw right?
>> In fact from the sip debug it looks like it does, but then I get the
>> dreaded "channel.c:5630 set_format: Unable to find a codec translation
>> path: (g729) -> (alaw)" and the call hangs up.  Why?
>>
>> Last minute thought: Is it possible that the caller is sending g729 in
>> RTP even though the SIP negotiation clearly chooses alaw?  Maybe I need
>> some RTP debugging.
>>
>> Asterisk 13.14.1 on Debian, using chan_sip.
>>
>> Here's the trace:
>>
>> <--- SIP read from UDP:SUPPLIER:5060 --->
>> INVITE sip:LOCAL at ASTERISK:5060 SIP/2.0
>> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9
>> From: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1
>> To: <sip:LOCAL at ASTERISK>
>> Call-ID: 205665777_90679951 at SUPPLIER
>> CSeq: 539098 INVITE
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
>> Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
>> Contact: <sip:REMOTE at SUPPLIER:5060>
>> P-Asserted-Identity: <sip:REMOTE at REMOTE-SUPPLIER;user=phone>
>> Supported: timer,100rel,precondition
>> Session-Expires: 1800
>> Min-SE: 90
>> Content-Length: 282
>> Content-Disposition: session; handling=required
>> Content-Type: application/sdp
>>
>> v=0
>> o=Sonus_UAC 176880 320591 IN IP4 SUPPLIER
>> s=SIP Media Capabilities
>> c=IN IP4 213.41.124.6
>> t=0 0
>> m=audio 8526 RTP/AVP 18 8 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sendrecv
>> a=ptime:20
>> <------------->
>> --- (17 headers 13 lines) ---
>> Sending to SUPPLIER:5060 (no NAT)
>> Sending to SUPPLIER:5060 (no NAT)
>> Using INVITE request as basis request - 205665777_90679951 at SUPPLIER
>> Found peer 'supplier' for 'REMOTE' from SUPPLIER:5060
>> Found RTP audio format 18
>> Found RTP audio format 8
>> Found RTP audio format 101
>> Found audio description format G729 for ID 18
>> Found audio description format PCMA for ID 8
>> Found audio description format telephone-event for ID 101
>> Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
>> Peer audio RTP is at port 213.41.124.6:8526
>> Looking for LOCAL in supplier-in (domain ASTERISK)
>> sip_route_dump: route/path hop: <sip:REMOTE at SUPPLIER:5060>
>>
>> So, all looking good here, we've worked out that the combined
>> capabilities are (alaw)
>>
>> <--- Transmitting (no NAT) to SUPPLIER:5060 --->
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER
>> From: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1
>> To: <sip:LOCAL at ASTERISK>
>> Call-ID: 205665777_90679951 at SUPPLIER
>> CSeq: 539098 INVITE
>> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
>> Supported: replaces, timer
>> Session-Expires: 1800;refresher=uas
>> Contact: <sip:LOCAL at ASTERISK:5060>
>> Content-Length: 0
>>
>>
>> <------------>
>>
>> <--- Transmitting (no NAT) to SUPPLIER:5060 --->
>> SIP/2.0 180 Ringing
>> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER
>> From: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1
>> To: <sip:LOCAL at ASTERISK>;tag=as4502927f
>> Call-ID: 205665777_90679951 at SUPPLIER
>> CSeq: 539098 INVITE
>> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
>> Supported: replaces, timer
>> Session-Expires: 1800;refresher=uas
>> Contact: <sip:LOCAL at ASTERISK:5060>
>> Content-Length: 0
>>
>>
>> <------------>
>> Audio is at 13948
>> Adding codec alaw to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>>
>> <--- Reliably Transmitting (no NAT) to SUPPLIER:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER
>> From: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1
>> To: <sip:LOCAL at ASTERISK>;tag=as4502927f
>> Call-ID: 205665777_90679951 at SUPPLIER
>> CSeq: 539098 INVITE
>> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
>> Supported: replaces, timer
>> Session-Expires: 1800;refresher=uas
>> Contact: <sip:LOCAL at ASTERISK:5060>
>> Content-Type: application/sdp
>> Require: timer
>> Content-Length: 264
>>
>> v=0
>> o=root 227409966 227409966 IN IP4 ASTERISK
>> s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
>> c=IN IP4 ASTERISK
>> t=0 0
>> m=audio 13948 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=maxptime:150
>> a=sendrecv
>>
>> <------------>
>>
>>
>> And that's good to, we've sent the OK for the INVITE saying that we want
>> alaw.
>>
>>
>> <--- SIP read from UDP:SUPPLIER:5060 --->
>> ACK sip:LOCAL at ASTERISK:5060 SIP/2.0
>> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5bc037285f864da9
>> From: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1
>> To: <sip:LOCAL at ASTERISK>;tag=as4502927f
>> Call-ID: 205665777_90679951 at SUPPLIER
>> CSeq: 539098 ACK
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> <------------->
>> --- (8 headers 0 lines) ---
>> [May 13 13:46:58] WARNING[7245][C-000031da]: channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)
>>
>> What's this nonsense!  Why is set_format trying to use g729!
>>
>> Scheduling destruction of SIP dialog '205665777_90679951 at SUPPLIER' in 32000 ms (Method: ACK)
>> set_destination: Parsing <sip:REMOTE at SUPPLIER:5060> for address/port to send to
>> set_destination: set destination to SUPPLIER:5060
>> Reliably Transmitting (no NAT) to SUPPLIER:5060:
>> BYE sip:REMOTE at SUPPLIER:5060 SIP/2.0
>> Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d
>> Max-Forwards: 70
>> From: <sip:LOCAL at ASTERISK>;tag=as4502927f
>> To: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1
>> Call-ID: 205665777_90679951 at SUPPLIER
>> CSeq: 102 BYE
>> User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
>> X-Asterisk-HangupCause: Normal Clearing
>> X-Asterisk-HangupCauseCode: 16
>> Content-Length: 0
>>
>>
>> ---
>>
>> <--- SIP read from UDP:SUPPLIER:5060 --->
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d
>> From: <sip:LOCAL at ASTERISK>;tag=as4502927f
>> To: <sip:REMOTE at SUPPLIER>;tag=gK02498cb1
>> Call-ID: 205665777_90679951 at SUPPLIER
>> CSeq: 102 BYE
>> Content-Length: 0
>>
>> <------------->
>> --- (7 headers 0 lines) ---
>> SIP Response message for INCOMING dialog BYE arrived
>> Really destroying SIP dialog '205665777_90679951 at SUPPLIER' Method: ACK
>>
>>
>> --
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