Hi,

We are trying to connect to IPV4 server from our pjsip ios app with NAT64.
We can able to register with the server successfully But when we make calls
their is no audio and call disconnects.
Below is the error.

19:39:43.256  pjsua_media.c  .....Call 1: updating media..

19:39:43.256  pjsua_media.c  ......pjmedia_stream_info_from_sdp() failed
for call_id 1 media 0: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......Error updating media call01:0:
Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......pjmedia_vid_stream_info_from_sdp()
failed for call_id 1 media 1: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......Error updating media call01:1:
Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_call.c  .....Unable to create media session: No
active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA)
[status=220048]

19:39:43.257  pjsua_core.c  ........TX 389 bytes Request msg
CANCEL/cseq=17639 (tdta0x1021ce200) to TCP 2001:2:0:1baa::4225:2ca6:5060:

What could be wrong? Can someone please help me on this.

Thanks,

Ashok Narvaneni.

Hi Ashok

Glad the module is more or less doing the job for you. As you can see in the README.md file on GitHub under bullet 2, there is an example on how you can activate this only when connecting to a IPv6 network. Then IPv4 continues to work as it has always done. Just hook it into the on_registration callback from pjsip and I think you will be fine.

Johan

From: pjsip on behalf of Ashok Narvaneni
Reply-To: pjsip list
Date: Tuesday, 4 April 2017 at 16:36
To: pjsip list
Subject: Re: [pjsip] NAT64 ios issue

Hi Colin,

Thanks again for your kind reply.
We will make the changes and post our findings.
Unfortunately this server is provided by our client which does't have ICE support.

Thanks,
Ashok Narvaneni.

On Tue, Apr 4, 2017 at 7:59 PM, Colin Morelli <colin.morelli@gmail.commailto:colin.morelli@gmail.com> wrote:
Ahsok,

This is one of the reasons I had written my own module (well, that and supporting ICE). The module I linked does not perform any checks about the current connection to the SIP server. Among other things, one of the additional things my modules does is only operate when connected to the signaling server via IPv6. This can be accomplished by a simple check in the on_rx and on_tx methods, before touching the SDP:

(tdata->tp_info.transport->factory->type & PJSIP_TRANSPORT_IPV6) == PJSIP_TRANSPORT_IPV6

With that check at the beginning of the module methods, it should simply ignore anything that happens on an IPv4 transport and leave the SDP in-tact.

As for PJSIP supporting dual stack mode - there's really no sense in trying to do this when ICE is available. PJSIP's ICE implementation can and does support dual stack. If you can enable ICE on your server side as well, that should work for your use case.

Best,
Colin

On Tue, Apr 4, 2017 at 10:23 AM, Ashok Narvaneni <ashoknarvaneni7@gmail.commailto:ashoknarvaneni7@gmail.com> wrote:
Hi Colin,

We integrated the module as you suggested and it is working fine.
However it breaks when we connect the device with an IPV4 network. We understand that pjsip does't support dual stack mode. So what is the best way to do make the App work with both IPV6 and IPV4 networks.
Please suggest...

Thanks,
Ashok Narvaneni.

On Fri, Mar 31, 2017 at 8:14 PM, Ashok Narvaneni <ashoknarvaneni7@gmail.commailto:ashoknarvaneni7@gmail.com> wrote:
Hi Colin,

Thank you very much for detailed answer, I will try that patch and post my findings.

Thanks,
Ashok.

On Fri, Mar 31, 2017 at 8:08 PM, Colin Morelli <colin.morelli@gmail.commailto:colin.morelli@gmail.com> wrote:
Ashok,

What's almost certainly happening here is that the SDP returned from your server contains only IPv4 media addresses. When PJSIP goes to compare the SDP to local capabilities, it finds only IPv4 addresses for the remote endpoint, but can't find any local IPv4 interfaces. As a result, it thinks there's no way it can communicate with the remote server and fails.

I have been able to fix this with two steps:

1. Enable ICE negotiation on both your media server and PJSIP client. ICE is helpful here because it allows the two endpoints to communicate multiple candidate media addresses. This step may be optional, although I've never really tested without this.
2. Write a PJSIP module that intercepts incoming/outgoing SDPs to work around the NAT64 issue. There's an example of one of these on githubhttps://github.com/johanlantz/pj-nat64, and I've written my own for my app (unfortunately can't share at this time). The general steps that need to be followed here are:

• Intercept on_tx and on_rx PJSIP events
• When sending an outgoing SDP, if signaling is connected over IPv6, then add an IPv4 address to the ICE candidates in the SDP (this step is optional, and depends on whether or not your server will explicitly reject an SDP that only contains IPv6 addresses)
• When receiving an incoming SDP, if signaling is connected over IPv6, then iterate the ICE candidates in the SDP, find all IPv4 addresses, synthesize IPv6 addresses for them (by passing the IPv4 addresses to pj_getaddrinfo, iOS will return a synthesized IPv6 address if connected to a NAT64 network), and add the synthesized IPv6 addresses to the ICE candidates list

As long as this module is registered with a higher priority than the PJSIP transports module, the SDP will be rewritten before pjmedia actually parses the SDP. By the time it gets to the media stack, it will see your synthesized IPv6 addresses, which it can support, and should be able to establish a media connection.

Couple of tips here: don't underestimate the nuances of this issue. It's not necessarily a hard problem, but there are a lot of cases to consider. Once you solve the first issue, you then have the issue of what happens when the user changes networks during a call (i.e. their phone switches wifi networks, or off of wifi entirely). Additionally, if you're working on Android, you probably have this issue on Android as well, but Google as far as I know does not test apps on a NAT64 network like Apple does.

Apologies I can't be more specific at this time but I hope this points you in the right direction. I think it'd be great for PJSIP to include this in the core. NAT64 networks are likely to become more prevalent, and it would be considerably less work to simply add to the core than it would be to maintain a separate module.

Best,
Colin

On Fri, Mar 31, 2017 at 10:14 AM, Ashok Narvaneni <ashoknarvaneni7@gmail.commailto:ashoknarvaneni7@gmail.com> wrote:
Hi,

We are trying to connect to IPV4 server from our pjsip ios app with NAT64.
We can able to register with the server successfully But when we make calls their is no audio and call disconnects.
Below is the error.

19:39:43.256  pjsua_media.c  .....Call 1: updating media..

19:39:43.256  pjsua_media.c  ......pjmedia_stream_info_from_sdp() failed for call_id 1 media 0: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......Error updating media call01:0: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......pjmedia_vid_stream_info_from_sdp() failed for call_id 1 media 1: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......Error updating media call01:1: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_call.c  .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048]

19:39:43.257  pjsua_core.c  ........TX 389 bytes Request msg CANCEL/cseq=17639 (tdta0x1021ce200) to TCP 2001:2:0:1baa::4225:2ca6:5060:

What could be wrong? Can someone please help me on this.

Thanks,

Ashok Narvaneni.

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Visit our blog: http://blog.pjsip.org

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For this problem I really do not know. I think our system does not pay that much attention to the Contact header but simply connects the registered device with the TCP/TLS socket used.

I doubt there is anything missing on the client side if you simply does not receive the INVITE. Are other packets reaching the device? MESSAGE, keep-alive, etc?

Johan

From: pjsip on behalf of Ashok Narvaneni
Reply-To: pjsip list
Date: Wednesday, 5 April 2017 at 14:40
To: pjsip list
Subject: Re: [pjsip] NAT64 ios issue

Thank you very much Colin and Johan for your help.

We have made some good progress. Below is the current status.
IPV4 - Phone register successfully, both Incoming and outgoing calls are working fine.
IPV6 - Phone register successfully, outgoing calls are working fine but when some one dial the extension of the phone under IPV6, FreeSwitch server doesn't send the INVITE to the phone .

Below is the register packet received on the server. Is there any other change we need to make on the app or Is that a FreeSwitch issue. Contact header have IPV6 address.

REGISTER sip:xxxxxx.xxx;transport=TCP SIP/2.0
Via: SIP/2.0/TCP [2001:2::aab1:dc80:49bb:fedc:3c1b]:63025;rport;branch=z9hG4bKPjq-wL2PpL4k0klpjECGxQ7duSoDQPzA6-;alias
Max-Forwards: 70
From: <sip:103@xxxxxx.xxxmailto:103@xxxxxx.xxx>;tag=3oiEgXukA1tcpLpzuDPslVusej23ql5a
To: <sip:103@xxxxxx.xxxmailto:103@xxxxxx.xxx>
Call-ID: rQmIvhvfa8krjErV-bKQTnDcsfK79INU
CSeq: 57007 REGISTER
User-Agent: SIPUA-1.0 iOS-10.0.1/arm-iPhone8,1/iOS-SDK
Supported: outbound, path
Contact: sip:103@[2001:2::aab1:dc80:49bb:fedc:3c1b]:63025;transport=TCP;ob;reg-id=1;+sip.instance="urn:uuid:00000000-0000-0000-0000-0000fea119f6"
Expires: 60
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

Thanks,
Ashok Narvaneni.

On Tue, Apr 4, 2017 at 8:15 PM, JOHAN LANTZ <johan.lantz@telefonica.commailto:johan.lantz@telefonica.com> wrote:
Hi Ashok

Glad the module is more or less doing the job for you. As you can see in the README.md file on GitHub under bullet 2, there is an example on how you can activate this only when connecting to a IPv6 network. Then IPv4 continues to work as it has always done. Just hook it into the on_registration callback from pjsip and I think you will be fine.

Johan

From: pjsip on behalf of Ashok Narvaneni
Reply-To: pjsip list
Date: Tuesday, 4 April 2017 at 16:36
To: pjsip list
Subject: Re: [pjsip] NAT64 ios issue

Hi Colin,

Thanks again for your kind reply.
We will make the changes and post our findings.
Unfortunately this server is provided by our client which does't have ICE support.

Thanks,
Ashok Narvaneni.

On Tue, Apr 4, 2017 at 7:59 PM, Colin Morelli <colin.morelli@gmail.commailto:colin.morelli@gmail.com> wrote:
Ahsok,

This is one of the reasons I had written my own module (well, that and supporting ICE). The module I linked does not perform any checks about the current connection to the SIP server. Among other things, one of the additional things my modules does is only operate when connected to the signaling server via IPv6. This can be accomplished by a simple check in the on_rx and on_tx methods, before touching the SDP:

(tdata->tp_info.transport->factory->type & PJSIP_TRANSPORT_IPV6) == PJSIP_TRANSPORT_IPV6

With that check at the beginning of the module methods, it should simply ignore anything that happens on an IPv4 transport and leave the SDP in-tact.

As for PJSIP supporting dual stack mode - there's really no sense in trying to do this when ICE is available. PJSIP's ICE implementation can and does support dual stack. If you can enable ICE on your server side as well, that should work for your use case.

Best,
Colin

On Tue, Apr 4, 2017 at 10:23 AM, Ashok Narvaneni <ashoknarvaneni7@gmail.commailto:ashoknarvaneni7@gmail.com> wrote:
Hi Colin,

We integrated the module as you suggested and it is working fine.
However it breaks when we connect the device with an IPV4 network. We understand that pjsip does't support dual stack mode. So what is the best way to do make the App work with both IPV6 and IPV4 networks.
Please suggest...

Thanks,
Ashok Narvaneni.

On Fri, Mar 31, 2017 at 8:14 PM, Ashok Narvaneni <ashoknarvaneni7@gmail.commailto:ashoknarvaneni7@gmail.com> wrote:
Hi Colin,

Thank you very much for detailed answer, I will try that patch and post my findings.

Thanks,
Ashok.

On Fri, Mar 31, 2017 at 8:08 PM, Colin Morelli <colin.morelli@gmail.commailto:colin.morelli@gmail.com> wrote:
Ashok,

What's almost certainly happening here is that the SDP returned from your server contains only IPv4 media addresses. When PJSIP goes to compare the SDP to local capabilities, it finds only IPv4 addresses for the remote endpoint, but can't find any local IPv4 interfaces. As a result, it thinks there's no way it can communicate with the remote server and fails.

I have been able to fix this with two steps:

1. Enable ICE negotiation on both your media server and PJSIP client. ICE is helpful here because it allows the two endpoints to communicate multiple candidate media addresses. This step may be optional, although I've never really tested without this.
2. Write a PJSIP module that intercepts incoming/outgoing SDPs to work around the NAT64 issue. There's an example of one of these on githubhttps://github.com/johanlantz/pj-nat64, and I've written my own for my app (unfortunately can't share at this time). The general steps that need to be followed here are:

• Intercept on_tx and on_rx PJSIP events
• When sending an outgoing SDP, if signaling is connected over IPv6, then add an IPv4 address to the ICE candidates in the SDP (this step is optional, and depends on whether or not your server will explicitly reject an SDP that only contains IPv6 addresses)
• When receiving an incoming SDP, if signaling is connected over IPv6, then iterate the ICE candidates in the SDP, find all IPv4 addresses, synthesize IPv6 addresses for them (by passing the IPv4 addresses to pj_getaddrinfo, iOS will return a synthesized IPv6 address if connected to a NAT64 network), and add the synthesized IPv6 addresses to the ICE candidates list

As long as this module is registered with a higher priority than the PJSIP transports module, the SDP will be rewritten before pjmedia actually parses the SDP. By the time it gets to the media stack, it will see your synthesized IPv6 addresses, which it can support, and should be able to establish a media connection.

Couple of tips here: don't underestimate the nuances of this issue. It's not necessarily a hard problem, but there are a lot of cases to consider. Once you solve the first issue, you then have the issue of what happens when the user changes networks during a call (i.e. their phone switches wifi networks, or off of wifi entirely). Additionally, if you're working on Android, you probably have this issue on Android as well, but Google as far as I know does not test apps on a NAT64 network like Apple does.

Apologies I can't be more specific at this time but I hope this points you in the right direction. I think it'd be great for PJSIP to include this in the core. NAT64 networks are likely to become more prevalent, and it would be considerably less work to simply add to the core than it would be to maintain a separate module.

Best,
Colin

On Fri, Mar 31, 2017 at 10:14 AM, Ashok Narvaneni <ashoknarvaneni7@gmail.commailto:ashoknarvaneni7@gmail.com> wrote:
Hi,

We are trying to connect to IPV4 server from our pjsip ios app with NAT64.
We can able to register with the server successfully But when we make calls their is no audio and call disconnects.
Below is the error.

19:39:43.256  pjsua_media.c  .....Call 1: updating media..

19:39:43.256  pjsua_media.c  ......pjmedia_stream_info_from_sdp() failed for call_id 1 media 0: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......Error updating media call01:0: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......pjmedia_vid_stream_info_from_sdp() failed for call_id 1 media 1: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_media.c  ......Error updating media call01:1: Unsupported address family (PJ_EAFNOTSUP)

19:39:43.257  pjsua_call.c  .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048]

19:39:43.257  pjsua_core.c  ........TX 389 bytes Request msg CANCEL/cseq=17639 (tdta0x1021ce200) to TCP 2001:2:0:1baa::4225:2ca6:5060:

What could be wrong? Can someone please help me on this.

Thanks,

Ashok Narvaneni.

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
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Hi Ashok

I think you simply have to dig into the code and fix this yourself. I only implemented this for the audio case so probably multiple media lines are not managed at all.

/Johan

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The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it.

Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

Hi Johan:

Trying to make video work with IPv6, and we see that crash is happening when the condition if (strlen(new_content_len_buf) <= current_content_len.slen) fails. It prints the log "Updated content length needs more bytes than old one, we must do expand and copy. TODO".  Could you please suggest ways of fixing this or elaborate on the log message "we must expand and copy"?

Any help with resolving this is highly appreciated.

Thank you,

Girish

From: pjsip pjsip-bounces@lists.pjsip.org on behalf of JOHAN LANTZ johan.lantz@telefonica.com
Sent: Tuesday, April 18, 2017 1:57 PM
To: pjsip list
Subject: Re: [pjsip] NAT64 ios issue

Hi Ashok

I think you simply have to dig into the code and fix this yourself. I only implemented this for the audio case so probably multiple media lines are not managed at all.

/Johan

Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción.

The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it.

Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

Could you send an example of the sdp you are processing?

I assume this simply means that previously your content length used for instance 3 chars, something like 980 and then after the manipulation content length is more than 1000 bytes meaning we need 4 characters instead of 3. IIRC my sdp´s were quite far away from the limit so I added that scenario as a corner case that I did not implement. I think you will be able to fix it yourself if you spend some time on it. I am going to spend a little time on this code this week but I can not promise I have time to add this scenario so I suggest you start by trying to fix it yourself (and contribute the fix back to GitHub if it works :-)

Johan

From: Girish Gopinath
Date: Wednesday, 10 May 2017 at 15:08
To: pjsip list, Johan Lantz
Subject: Re: [pjsip] NAT64 ios issue

Updated content length needs more bytes than old one, we must do expand

Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción.

The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it.

Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

Hi Johan:

Thank you for the reply.  Added logs below from an attempt to switch from audio call to video. The content length is never more than 1000, but in the logs it says 1253. Looks like I'm missing something here.

Thank you,

Girish

15:20:08.575  pjsua_core.c  ....TX 1398 bytes Request msg INVITE/cseq=11944 (tdta0x12284d200) to TCP 2001:2:0:1baa::4225:2ca6:5060:
INVITE sip:mod_sofia@[2001:2:0:1baa::4225:2ca6]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP XX.XX.117.54:47919;rport;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias
Max-Forwards: 70
From: sip:104@XX.XX.117.54;ob;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB
To: "SipApp100" sip:103@devpbx.pbxonline.com;tag=5Zv34Z5NQ22QF
Contact: sip:104@[2001:2::aab1:292f:d2ac:6e72:35f5]:58184;transport=TCP;ob
Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f
CSeq: 11944 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: VoIPUA-1.0 iOS-10.3.1/arm-iPhone8,1/iOS-SDK
Content-Type: application/sdp
Content-Length:  668

v=0
o=- 3703398596 3703398599 IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 0 99 9 96
c=IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5
b=TIAS:64000
a=rtcp:4001 IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:99 speex/32000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
m=video 4004 RTP/AVP 97
c=IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5
b=TIAS:256000
a=rtcp:4005 IN IP6 2001:2::aab1:292f:d2ac:6e72:35f5
a=sendrecv
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1

--end msg--
15:20:08.781    pj_nat64.c !.ipv6_mod_on_rx
15:20:08.781    pj_nat64.c  .Incoming INVITE or 200 OK. If they contain IPv4 addresses, we need to change to ipv6
15:20:08.781    pj_nat64.c  .Incoming INVITE or 200 with IPv4 address***
15:20:08.781    pj_nat64.c  .SIP/2.0 100 Trying
Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias
From: sip:104@XX.XX.117.54;ob;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB
To: "SipApp100" sip:103@devpbx.pbxonline.com;tag=5Zv34Z5NQ22QF
Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f
CSeq: 11944 INVITE
User-Agent: Fusion-PBX
Content-Length: 0

15:20:08.781    pj_nat64.c  .***************************************************************
15:20:08.781    pj_nat64.c  .Scanner syntax error at
15:20:08.781    pj_nat64.c  .Error: Parsing of the incoming INVITE/200 OK failed at . Leave incoming buffer as is
15:20:08.781  pjsua_core.c  .RX 375 bytes Response msg 100/INVITE/cseq=11944 (rdata0x122890f30) from TCP 2001:2:0:1baa::4225:2ca6:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias
From: sip:104@XX.XX.117.54;ob;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB
To: "SipApp100" sip:103@devpbx.pbxonline.com;tag=5Zv34Z5NQ22QF
Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f
CSeq: 11944 INVITE
User-Agent: Fusion-PBX
Content-Length: 0

--end msg--
15:20:08.789    pj_nat64.c  .ipv6_mod_on_rx
15:20:08.789    pj_nat64.c  .Incoming INVITE or 200 OK. If they contain IPv4 addresses, we need to change to ipv6
15:20:08.789    pj_nat64.c  .Incoming INVITE or 200 with IPv4 address***
15:20:08.789    pj_nat64.c  .SIP/2.0 200 OK
Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias
From: sip:104@XX.XX.117.54;ob;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB
To: "SipApp100" sip:103@devpbx.pbxonline.com;tag=5Zv34Z5NQ22QF
Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f
CSeq: 11944 INVITE
Contact: sip:mod_sofia@XX.XX.44.166:5060;transport=tcp
User-Agent: Fusion-PBX
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 358

v=0
o=FreeSWITCH 1494392300 1494392302 IN IP4 XX.XX.44.166
s=FreeSWITCH
c=IN IP4 XX.XX.44.166
t=0 0
m=audio 16812 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=rtcp:16813 IN IP4 XX.XX.44.166
m=video 23982 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1

15:20:08.789    pj_nat64.c  .***************************************************************
15:20:08.789    pj_nat64.c  .Extracted ip4 address as XX.XX.44.166
15:20:08.793    pj_nat64.c  .Extracted ip4 address as XX.XX.44.166
15:20:08.794    pj_nat64.c  .Extracted ip4 address as XX.XX.44.166
15:20:08.795    pj_nat64.c  .Scanner syntax error at
15:20:08.795    pj_nat64.c  .Current Content-Length is: 358 and new Content-Length is 1253 .
15:20:08.795    pj_nat64.c  .Updated content length needs more bytes than old one, we must do expand and copy. TODO
15:20:08.795    pj_nat64.c  .We have successfully parsed the INVITE/200 OK until EOF. Replace rx buffer. pjsip will now print the modified rx packet.
15:20:08.795    pj_nat64.c  .Host in Contact header is XX.XX.44.166
15:20:08.796  pjsua_core.c  .RX 1914 bytes Response msg 200/INVITE/cseq=11944 (rdata0x122890f30) from TCP 2001:2:0:1baa::4225:2ca6:5060:
SIP/2.0 200 OK
Via: SIP/2.0/TCP XX.XX.117.54:47919;rport=47919;branch=z9hG4bKPjhfiX5T7SMfffGsyd8n6wdmhaKL2Yk263;alias
From: sip:104@XX.XX.117.54;ob;tag=YuPVVYlB3CRimhCfi-R9-rriu8ba5MbB
To: "SipApp100" sip:103@devpbx.pbxonline.com;tag=5Zv34Z5NQ22QF
Call-ID: ca1ba4ba-b008-1235-8fb0-00163e70e05f
CSeq: 11944 INVITE
Contact: sip:mod_sofia@XX.XX.44.166:5060;transport=tcp
User-Agent: Fusion-PBX
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 358

v=0
o=FreeSWITCH 1494392300 1494392302 IN IP6 2001:2:0:1baa::4225:2ca6
s=FreeSWITCH
c=IN IP6 2001:2:0:1baa::4225:2ca6
t=0 0
m=audio 16812 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=rtcp:16813 IN IP6 2001:2:0:1baa::4225:2ca6
m=video 23982 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1

--end msg--

15:20:08.797          sdp.c  ....Error parsing SDP in line 15 col 0: Success
Assertion failed: (ctx.last_error != PJ_SUCCESS), function pjmedia_sdp_parse, file ../src/pjmedia/sdp.c, line 1349.
(lldb)

From: JOHAN LANTZ johan.lantz@telefonica.com
Sent: Wednesday, May 10, 2017 7:00 PM
To: Girish Gopinath; pjsip list
Subject: Re: [pjsip] NAT64 ios issue

Could you send an example of the sdp you are processing?

I assume this simply means that previously your content length used for instance 3 chars, something like 980 and then after the manipulation content length is more than 1000 bytes meaning we need 4 characters instead of 3. IIRC my sdp´s were quite far away from the limit so I added that scenario as a corner case that I did not implement. I think you will be able to fix it yourself if you spend some time on it. I am going to spend a little time on this code this week but I can not promise I have time to add this scenario so I suggest you start by trying to fix it yourself (and contribute the fix back to GitHub if it works :-)

Johan

From: Girish Gopinath
Date: Wednesday, 10 May 2017 at 15:08
To: pjsip list, Johan Lantz
Subject: Re: [pjsip] NAT64 ios issue

Updated content length needs more bytes than old one, we must do expand

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